draft-ietf-avtcore-multi-party-rtt-mix-20.txt   rfc9071.txt 
AVTCore G. Hellstrom Internet Engineering Task Force (IETF) G. Hellström
Internet-Draft Gunnar Hellstrom Accessible Communication Request for Comments: 9071 GHAccess
Updates: 4103 (if approved) 26 May 2021 Updates: 4103 July 2021
Intended status: Standards Track Category: Standards Track
Expires: 27 November 2021 ISSN: 2070-1721
RTP-mixer formatting of multiparty Real-time text RTP-Mixer Formatting of Multiparty Real-Time Text
draft-ietf-avtcore-multi-party-rtt-mix-20
Abstract Abstract
This document provides enhancements for RFC 4103 real-time text This document provides enhancements of real-time text (as specified
mixing suitable for a centralized conference model that enables in RFC 4103) suitable for mixing in a centralized conference model,
source identification and rapidly interleaved transmission of text enabling source identification and rapidly interleaved transmission
from different sources. The intended use is for real-time text of text from different sources. The intended use is for real-time
mixers and participant endpoints capable of providing an efficient text mixers and participant endpoints capable of providing an
presentation or other treatment of a multiparty real-time text efficient presentation or other treatment of a multiparty real-time
session. The specified mechanism builds on the standard use of the text session. The specified mechanism builds on the standard use of
Contributing Source (CSRC) list in the Realtime Protocol (RTP) packet the Contributing Source (CSRC) list in the Real-time Transport
for source identification. The method makes use of the same "text/ Protocol (RTP) packet for source identification. The method makes
t140" and "text/red" formats as for two-party sessions. use of the same "text/t140" and "text/red" formats as for two-party
sessions.
Solutions using multiple RTP streams in the same RTP session are Solutions using multiple RTP streams in the same RTP session are
briefly mentioned, as they could have some benefits over the RTP- briefly mentioned, as they could have some benefits over the RTP-
mixer model. The possibility to implement the solution in a wide mixer model. The RTP-mixer model was selected to be used for the
range of existing RTP implementations made the RTP-mixer model be fully specified solution in this document because it can be applied
selected to be fully specified in this document. to a wide range of existing RTP implementations.
A capability exchange is specified so that it can be verified that a A capability exchange is specified so that it can be verified that a
mixer and a participant can handle the multiparty-coded real-time mixer and a participant can handle the multiparty-coded real-time
text stream using the RTP-mixer method. The capability is indicated text stream using the RTP-mixer method. The capability is indicated
by use of an RFC 8866 Session Description Protocol (SDP) media by the use of a Session Description Protocol (SDP) (RFC 8866) media
attribute "rtt-mixer". attribute, "rtt-mixer".
The document updates RFC 4103 "RTP Payload for Text Conversation". This document updates RFC 4103 ("RTP Payload for Text Conversation").
A specification of how a mixer can format text for the case when the A specification for how a mixer can format text for the case when the
endpoint is not multiparty-aware is also provided. endpoint is not multiparty aware is also provided.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This is an Internet Standards Track document.
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
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working documents as Internet-Drafts. The list of current Internet-
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Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
material or to cite them other than as "work in progress." Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
This Internet-Draft will expire on 27 November 2021. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc9071.
Copyright Notice Copyright Notice
Copyright (c) 2021 IETF Trust and the persons identified as the Copyright (c) 2021 IETF Trust and the persons identified as the
document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 4 1. Introduction
1.1. Terminology . . . . . . . . . . . . . . . . . . . . . . . 6 1.1. Terminology
1.2. Selected solution and considered alternatives . . . . . . 7 1.2. Main Method, Fallback Method, and Considered Alternatives
1.3. Intended application . . . . . . . . . . . . . . . . . . 9 1.3. Intended Application
2. Overview of the two specified solutions and selection of 2. Overview of the Two Specified Solutions and Selection of Method
method . . . . . . . . . . . . . . . . . . . . . . . . . 10 2.1. The RTP-Mixer-Based Solution for Multiparty-Aware Endpoints
2.1. The RTP-mixer-based solution for multiparty-aware 2.2. Mixing for Multiparty-Unaware Endpoints
endpoints . . . . . . . . . . . . . . . . . . . . . . . . 10 2.3. Offer/Answer Considerations
2.2. Mixing for multiparty-unaware endpoints . . . . . . . . . 11 2.4. Actions Depending on Capability Negotiation Result
2.3. Offer/answer considerations . . . . . . . . . . . . . . . 11 3. Details for the RTP-Mixer-Based Mixing Method for
2.4. Actions depending on capability negotiation result . . . 13 Multiparty-Aware Endpoints
3. Details for the RTP-mixer-based mixing method for 3.1. Use of Fields in the RTP Packets
multiparty-aware endpoints . . . . . . . . . . . . . . . 13 3.2. Initial Transmission of a BOM Character
3.1. Use of fields in the RTP packets . . . . . . . . . . . . 13 3.3. Keep-Alive
3.2. Initial transmission of a BOM character . . . . . . . . . 14 3.4. Transmission Interval
3.3. Keep-alive . . . . . . . . . . . . . . . . . . . . . . . 14 3.5. Only One Source per Packet
3.4. Transmission interval . . . . . . . . . . . . . . . . . . 14 3.6. Do Not Send Received Text to the Originating Source
3.5. Only one source per packet . . . . . . . . . . . . . . . 15 3.7. Clean Incoming Text
3.6. Do not send received text to the originating source . . . 15 3.8. Principles of Redundant Transmission
3.7. Clean incoming text . . . . . . . . . . . . . . . . . . . 16 3.9. Text Placement in Packets
3.8. Redundant transmission principles . . . . . . . . . . . . 16 3.10. Empty T140blocks
3.9. Text placement in packets . . . . . . . . . . . . . . . . 16 3.11. Creation of the Redundancy
3.10. Empty T140blocks . . . . . . . . . . . . . . . . . . . . 17 3.12. Timer Offset Fields
3.11. Creation of the redundancy . . . . . . . . . . . . . . . 17 3.13. Other RTP Header Fields
3.12. Timer offset fields . . . . . . . . . . . . . . . . . . . 18 3.14. Pause in Transmission
3.13. Other RTP header fields . . . . . . . . . . . . . . . . . 18 3.15. RTCP Considerations
3.14. Pause in transmission . . . . . . . . . . . . . . . . . . 18 3.16. Reception of Multiparty Contents
3.15. RTCP considerations . . . . . . . . . . . . . . . . . . . 19 3.17. Performance Considerations
3.16. Reception of multiparty contents . . . . . . . . . . . . 19 3.18. Security for Session Control and Media
3.17. Performance considerations . . . . . . . . . . . . . . . 21 3.19. SDP Offer/Answer Examples
3.18. Security for session control and media . . . . . . . . . 21 3.20. Packet Sequence Example from Interleaved Transmission
3.19. SDP offer/answer examples . . . . . . . . . . . . . . . . 22 3.21. Maximum Character Rate "cps" Setting
3.20. Packet sequence example from interleaved transmission . . 23 4. Presentation-Level Considerations
3.21. Maximum character rate "cps" . . . . . . . . . . . . . . 26 4.1. Presentation by Multiparty-Aware Endpoints
4. Presentation level considerations . . . . . . . . . . . . . . 26 4.2. Multiparty Mixing for Multiparty-Unaware Endpoints
4.1. Presentation by multiparty-aware endpoints . . . . . . . 27 5. Relationship to Conference Control
4.2. Multiparty mixing for multiparty-unaware endpoints . . . 29 5.1. Use with SIP Centralized Conferencing Framework
5. Relation to Conference Control . . . . . . . . . . . . . . . 35 5.2. Conference Control
5.1. Use with SIP centralized conferencing framework . . . . . 36 6. Gateway Considerations
5.2. Conference control . . . . . . . . . . . . . . . . . . . 36 6.1. Gateway Considerations with Textphones
6. Gateway Considerations . . . . . . . . . . . . . . . . . . . 36 6.2. Gateway Considerations with WebRTC
6.1. Gateway considerations with Textphones . . . . . . . . . 36 7. Updates to RFC 4103
6.2. Gateway considerations with WebRTC . . . . . . . . . . . 36 8. Congestion Considerations
7. Updates to RFC 4103 . . . . . . . . . . . . . . . . . . . . . 37 9. IANA Considerations
8. Congestion considerations . . . . . . . . . . . . . . . . . . 38 9.1. Registration of the "rtt-mixer" SDP Media Attribute
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 38 10. Security Considerations
9.1. Registration of the "rtt-mixer" SDP media attribute . . . 38 11. References
10. Security Considerations . . . . . . . . . . . . . . . . . . . 39 11.1. Normative References
11. Change history . . . . . . . . . . . . . . . . . . . . . . . 40 11.2. Informative References
11.1. Changes included in Acknowledgements
draft-ietf-avtcore-multi-party-rtt-mix-20 . . . . . . . 40 Author's Address
11.2. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-19 . . . . . . . 40
11.3. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-18 . . . . . . . 40
11.4. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-17 . . . . . . . 40
11.5. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-16 . . . . . . . 40
11.6. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-15 . . . . . . . 41
11.7. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-14 . . . . . . . 41
11.8. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-13 . . . . . . . 41
11.9. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-12 . . . . . . . 42
11.10. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-11 . . . . . . . 42
11.11. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-10 . . . . . . . 42
11.12. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-09 . . . . . . . 42
11.13. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-08 . . . . . . . 43
11.14. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-07 . . . . . . . 43
11.15. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-06 . . . . . . . 43
11.16. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-05 . . . . . . . 43
11.17. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-04 . . . . . . . 43
11.18. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-03 . . . . . . . 44
11.19. Changes included in
draft-ietf-avtcore-multi-party-rtt-mix-02 . . . . . . . 45
11.20. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01 . . 45
11.21. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-03 to
draft-ietf-avtcore-multi-party-rtt-mix-00 . . . . . . . 45
11.22. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-02 to
-03 . . . . . . . . . . . . . . . . . . . . . . . . . . 45
11.23. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-01 to
-02 . . . . . . . . . . . . . . . . . . . . . . . . . . 46
11.24. Changes from
draft-hellstrom-avtcore-multi-party-rtt-source-00 to
-01 . . . . . . . . . . . . . . . . . . . . . . . . . . 47
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 47
12.1. Normative References . . . . . . . . . . . . . . . . . . 47
12.2. Informative References . . . . . . . . . . . . . . . . . 48
Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 49
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 49
1. Introduction 1. Introduction
"RTP Payload for Text Conversation" [RFC4103] specifies use of the "RTP Payload for Text Conversation" [RFC4103] specifies the use of
Real-Time Transport Protocol (RTP) [RFC3550] for transmission of the Real-time Transport Protocol (RTP) [RFC3550] for transmission of
real-time text (RTT) and the "text/t140" format. It also specifies a real-time text (often called RTT) and the "text/t140" format. It
redundancy format "text/red" for increased robustness. The "text/ also specifies a redundancy format, "text/red", for increased
red" format is registered in [RFC4102]. robustness. The "text/red" format is registered in [RFC4102].
Real-time text is usually provided together with audio and sometimes Real-time text is usually provided together with audio and sometimes
with video in conversational sessions. with video in conversational sessions.
A requirement related to multiparty sessions from the presentation A requirement related to multiparty sessions from the presentation-
level standard T.140 [T140] for real-time text is: "The display of level standard T.140 [T140] for real-time text is as follows:
text from the members of the conversation should be arranged so that
the text from each participant is clearly readable, and its source | The display of text from the members of the conversation should be
and the relative timing of entered text is visualized in the | arranged so that the text from each participant is clearly
display." | readable, and its source and the relative timing of entered text
| is visualized in the display.
Another requirement is that the mixing procedure must not introduce Another requirement is that the mixing procedure must not introduce
delays in the text streams that are experienced to be disturbing the delays in the text streams that could be perceived as disruptive to
real-time experience of the receiving users. the real-time experience of the receiving users.
Use of RTT is increasing, and specifically, use in emergency calls is The use of real-time text is increasing, and specifically, use in
increasing. Emergency call use requires multiparty mixing because it emergency calls is increasing. Emergency call use requires
is common that one agent needs to transfer the call to another multiparty mixing, because it is common that one agent needs to
specialized agent but is obliged to stay on the call at least to transfer the call to another specialized agent but is obliged to stay
verify that the transfer was successful. Mixer implementations for on the call to at least verify that the transfer was successful.
RFC 4103 "RTP Payload for Text Conversation" can use traditional RFC Mixer implementations for RFC 4103 ("RTP Payload for Text
3550 RTP functions for mixing and source identification, but the Conversation") can use traditional RTP functions (RFC 3550) for
performance of the mixer when giving turns for the different sources mixing and source identification, but the performance of the mixer
to transmit is limited when using the default transmission when giving turns for the different sources to transmit is limited
characteristics with redundancy. when using the default transmission characteristics with redundancy.
The redundancy scheme of [RFC4103] enables efficient transmission of The redundancy scheme described in [RFC4103] enables efficient
earlier transmitted redundant text in packets together with new text. transmission of earlier transmitted redundant text in packets
However, the redundancy header format has no source indicators for together with new text. However, the redundancy header format has no
the redundant transmissions. The redundant parts in a packet must source indicators for the redundant transmissions. The redundant
therefore be from the same source as the new text. The recommended parts in a packet must therefore be from the same source as the new
transmission is one new and two redundant generations of text text. The recommended transmission is one new and two redundant
(T140blocks) in each packet and the recommended transmission interval generations of text (T140blocks) in each packet, and the recommended
for two-party use is 300 ms. transmission interval for two-party use is 300 ms.
Real-time text mixers for multiparty sessions need to include the Real-time text mixers for multiparty sessions need to include the
source with each transmitted group of text from a conference source with each transmitted group of text from a conference
participant so that the text can be transmitted interleaved with text participant so that the text can be transmitted interleaved with text
groups from different sources at the rate they are created. This groups from different sources at the rate at which they are created.
enables the text groups to be presented by endpoints in suitable This enables the text groups to be presented by endpoints in a
grouping with other text from the same source. suitable grouping with other text from the same source.
The presentation can then be arranged so that text from different The presentation can then be arranged so that text from different
sources can be presented in real-time and easily read. At the same sources can be presented in real time and easily read. At the same
time it is possible for a reading user to perceive approximately when time, it is possible for a reading user to perceive approximately
the text was created in real time by the different parties. The when the text was created in real time by the different parties. The
transmission and mixing is intended to be done in a general way, so transmission and mixing are intended to be done in a general way, so
that presentation can be arranged in a layout decided by the that presentation can be arranged in a layout decided upon by the
endpoint. receiving endpoint.
There are existing implementations of RFC 4103 in endpoints without Existing implementations of RFC 4103 in endpoints that do not
the updates from this document. These will not be able to receive implement the updates specified in this document cannot be expected
and present real-time text mixed for multiparty-aware endpoints. to properly present real-time text mixed for multiparty-aware
endpoints.
A negotiation mechanism is therefore needed for verification if the A negotiation mechanism is therefore needed to verify if the parties
parties are able to handle a common method for multiparty (1) are able to handle a common method for multiparty transmissions
transmission and agreeing on using that method. and (2) can agree on using that method.
A fallback mixing procedure is also needed for cases when the A fallback mixing procedure is also needed for cases when the
negotiation result indicates that a receiving endpoint is not capable negotiation result indicates that a receiving endpoint is not capable
of handling the mixed format. Multiparty-unaware endpoints would of handling the mixed format. Multiparty-unaware endpoints would
possibly otherwise present all received multiparty mixed text as if possibly otherwise present all received multiparty mixed text as if
it came from the same source regardless of any accompanying source it came from the same source regardless of any accompanying source
indication coded in fields in the packet. Or they may have other indication coded in fields in the packet. Or, they may have other
undesirable ways of acting on the multiparty content. The fallback undesirable ways of acting on the multiparty content. The fallback
method is called the mixing procedure for multiparty-unaware method is called the mixing procedure for multiparty-unaware
endpoints. The fallback method is naturally not expected to meet all endpoints. The fallback method is naturally not expected to meet all
performance requirements placed on the mixing procedure for performance requirements placed on the mixing procedure for
multiparty-aware endpoints. multiparty-aware endpoints.
The document updates [RFC4103] by introducing an attribute for This document updates [RFC4103] by introducing an attribute for
declaring support of the RTP-mixer-based multiparty mixing case and declaring support of the RTP-mixer-based multiparty-mixing case and
rules for source indications and interleaving of text from different rules for source indications and interleaving of text from different
sources. sources.
1.1. Terminology 1.1. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in BCP "OPTIONAL" in this document are to be interpreted as described in
14 [RFC2119] [RFC8174] when, and only when, they appear in all BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown above. capitals, as shown here.
The terms Source Description (SDES), Canonical name (CNAME), Name The terms "Source Description" (SDES), "Canonical Name" (CNAME),
(NAME), Synchronization Source (SSRC), Contributing Source (CSRC), "Name" (NAME), "Synchronization Source" (SSRC), "Contributing Source"
CSRC list, CSRC count [CC], Real-Time control protocol (RTCP), RTP- (CSRC), "CSRC list", "CSRC count" (CC), "RTP Control Protocol"
mixer, RTP-translator are defined in [RFC3550]. (RTCP), and "RTP mixer" are defined in [RFC3550].
"real-time text" (RTT) is text transmitted instantly as it is typed
or created. Recipients can immediately read the message while it is
being written, without waiting.
The term "T140block" is defined in [RFC4103] to contain one or more The term "T140block" is defined in [RFC4103] to contain one or more
T.140 code elements. T.140 code elements.
"TTY" stands for a textphone type used in North America. "TTY" stands for a textphone type used in North America.
Web based real-time communication (WebRTC) is specified by the World Web Real-Time Communication (WebRTC) is specified by the World Wide
Wide Web Consortium (W3C) and IETF. See [RFC8825]. Web Consortium (W3C) and the IETF. See [RFC8825].
"DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension "DTLS-SRTP" is a Datagram Transport Layer Security (DTLS) extension
for use with Secure Real-Time Transport Protocol/Secure Real-Time for use with the Secure Real-time Transport Protocol / Secure Real-
Control Protocol (SRTP/SRTCP) specified in [RFC5764]. time Transport Control Protocol (SRTP/SRTCP) as specified in
[RFC5764].
"multiparty-aware" describes an endpoint receiving real-time text The term "multiparty aware" describes an endpoint that (1) receives
from multiple sources through a common conference mixer being able to real-time text from multiple sources through a common conference
present the text in real-time, separated by source, and presented so mixer, (2) is able to present the text in real time, separated by
that a user can get an impression of the approximate relative timing source, and (3) presents the text so that a user can get an
of text from different parties. impression of the approximate relative timing of text from different
parties.
"multiparty-unaware" describes an endpoint not itself being able to The term "multiparty unaware" describes an endpoint that cannot
separate text from different sources when received through a common itself separate text from different sources when the text is received
conference mixer. through a common conference mixer.
1.2. Selected solution and considered alternatives 1.2. Main Method, Fallback Method, and Considered Alternatives
A number of alternatives were considered when searching an efficient A number of alternatives were considered when searching for an
and easily implemented multiparty method for real-time text. This efficient and easily implemented multiparty method for real-time
section explains a few of them briefly. text. This section briefly explains a few of them.
Multiple RTP streams, one per participant Multiple RTP streams, one per participant:
One RTP stream per source would be sent in the same RTP session One RTP stream per source would be sent in the same RTP session
with the "text/red" format. From some points of view, use of with the "text/red" format. From some points of view, the use of
multiple RTP streams, one for each source, sent in the same RTP multiple RTP streams, one for each source, sent in the same RTP
session would be efficient, and would use exactly the same packet session would be efficient and would use exactly the same packet
format as [RFC4103] and the same payload type. A couple of format as [RFC4103] and the same payload type. A couple of
relevant scenarios using multiple RTP-streams are specified in relevant scenarios using multiple RTP streams are specified in
"RTP Topologies" [RFC7667]. One possibility of special interest "RTP Topologies" [RFC7667]. One possibility of special interest
is the Selective Forwarding Middlebox (SFM) topology specified in is the Selective Forwarding Middlebox (SFM) topology specified in
RFC 7667 section 3.7 that could enable end-to-end encryption. In Section 3.7 of [RFC7667], which could enable end-to-end
contrast to audio and video, real-time text is only transmitted encryption. In contrast to audio and video, real-time text is
when the users actually transmit information. Thus, an SFM only transmitted when the users actually transmit information.
solution would not need to exclude any party from transmission Thus, an SFM solution would not need to exclude any party from
under normal conditions. In order to allow the mixer to convey transmission under normal conditions. In order to allow the mixer
the packets with the payload preserved and encrypted, an SFM to convey the packets with the payload preserved and encrypted, an
solution would need to act on some specific characteristics of the SFM solution would need to act on some specific characteristics of
"text/red" format. The redundancy headers are part of the the "text/red" format. The redundancy headers are part of the
payload, so the receiver would need to just assume that the payload, so the receiver would need to just assume that the
payload type number in the redundancy header is for "text/t140". payload type number in the redundancy header is for "text/t140".
The characters per second parameter (cps) would need to act per The characters per second ("cps") parameter would need to act per
stream. The relation between the SSRC and the source would need stream. The relationship between the SSRC and the source would
to be conveyed in some specified way, e.g., in the CSRC. Recovery need to be conveyed in some specified way, e.g., in the CSRC.
and loss detection would preferably be based on sequence number Recovery and loss detection would preferably be based on RTP
gap detection. Thus, sequence number gaps in the incoming stream sequence number gap detection. Thus, sequence number gaps in the
to the mixer would need to be reflected in the stream to the incoming stream to the mixer would need to be reflected in the
participant, with no new gaps created by the mixer. However, the stream to the participant, with no new gaps created by the mixer.
RTP implementation in both mixers and endpoints need to support However, the RTP implementation in both mixers and endpoints needs
multiple streams in the same RTP session in order to use this to support multiple streams in the same RTP session in order to
mechanism. For best deployment opportunity, it should be possible use this mechanism. To provide the best opportunities for
to upgrade existing endpoint solutions to be multiparty-aware with deployment, it should be possible to upgrade existing endpoint
a reasonable effort. There is currently a lack of support for solutions to be multiparty aware with a reasonable amount of
multi-stream RTP in certain implementations. This fact led to effort. There is currently a lack of support for multi-stream RTP
this solution being only briefly mentioned in this document as an in certain implementations. This fact led to only brief mention
option for further study. of this solution in this document as an option for further study.
RTP-mixer-based method for multiparty-aware endpoints RTP-mixer-based method for multiparty-aware endpoints:
The "text/red" format in RFC 4103 is sent with a shorter The "text/red" format as defined in RFC 4102 and applied in RFC
transmission interval with the RTP-mixer method and indicating the 4103 is sent with the RTP-mixer method indicating the source in
source in the CSRC field. The "text/red" format with a "text/ the CSRC field. The "text/red" format with a "text/t140" payload
t140" payload in a single RTP stream can be sent when text is in a single RTP stream can be sent when text is available from the
available from the call participants instead of at the regular 300 call participants instead of at the regular 300 ms intervals.
ms. Transmission of packets with text from different sources can Transmission of packets with text from different sources can then
then be done smoothly while simultaneous transmission occurs as be done smoothly while simultaneous transmission occurs as long as
long as it is not limited by the maximum character rate "cps". it is not limited by the maximum character rate "cps" value. With
With ten participants sending text simultaneously, the switching ten participants sending text simultaneously, the switching and
and transmission performance is good. With more simultaneously transmission performance is good. With more simultaneously
sending participants, and with receivers having the default sending participants and with receivers at default capacity, there
capacity there will be a noticeable jerkiness and delay in text will be a noticeable jerkiness and delay in text presentation.
presentation. The jerkiness will be more expressed the more The more participants who send text simultaneously, the more
participants who send text simultaneously. Two seconds jerkiness jerkiness will occur. Two seconds of jerkiness will be noticeable
will be noticeable and slightly unpleasant, but it corresponds in and slightly unpleasant, but it corresponds in time to what typing
time to what typing humans often cause by hesitation or changing humans often cause by hesitating or changing position while
position while typing. A benefit of this method is that no new typing. A benefit of this method is that no new packet format
packet format needs to be introduced and implemented. Since needs to be introduced and implemented. Since simultaneous typing
simultaneous typing by more than two parties is expected to be by more than two parties is expected to be very rare -- as
very rare as described in Section 1.3, this method can be used described in Section 1.3 -- this method can be used successfully
successfully with good performance. Recovery of text in case of with good performance. Recovery of text in the case of packet
packet loss is based on analysis of timestamps of received loss is based on analysis of timestamps of received redundancy
redundancy versus earlier received text. Negotiation is based on versus earlier received text. Negotiation is based on a new SDP
a new SDP media attribute "rtt-mixer". This method is selected to media attribute, "rtt-mixer". This method was selected to be the
be the main one specified in this document. main method specified in this document.
Multiple sources per packet Multiple sources per packet:
A new "text" media subtype would be specified with up to 15 A new "text" media subtype would be specified with up to 15
sources in each packet. The mechanism would make use of the RTP sources in each packet. The mechanism would make use of the RTP-
mixer model specified in RTP [RFC3550]. The sources are indicated mixer model specified in RTP [RFC3550]. The sources would be
in strict order in the CSRC list of the RTP packets. The CSRC indicated in strict order in the CSRC list of the RTP packets.
list can have up to 15 members. Therefore, text from up to 15 The CSRC list can have up to 15 members. Therefore, text from up
sources can be included in each packet. Packets are normally sent to 15 sources can be included in each packet. Packets are
with 300 ms intervals. The mean delay will be 150 ms. A new normally sent at 300 ms intervals. The mean delay would be 150
redundancy packet format is specified. This method would result ms. A new redundancy packet format would be specified. This
in good performance, but would require standardization and method would result in good performance but would require
implementation of new releases in the target technologies that standardization and implementation of new releases in the target
would take more time than desirable to complete. It was therefore technologies; these would take more time than desirable to
not selected to be included in this document. complete. It was therefore not selected to be included in this
document.
Mixing for multiparty-unaware endpoints Mixing for multiparty-unaware endpoints:
Presentation of text from multiple parties is prepared by the The presentation of text from multiple parties is prepared by the
mixer in one single stream. It is desirable to have a method that mixer in one single stream. It is desirable to have a method that
does not require any modifications in existing user devices does not require any modifications in existing user devices
implementing RFC 4103 for RTT without explicit support of implementing RFC 4103 for real-time text without explicit support
multiparty sessions. This is possible by having the mixer insert of multiparty sessions. This is made possible by having the mixer
a new line and a text formatted source label before each switch of insert a new line and a text-formatted source label before each
text source in the stream. Switch of source can only be done in switch of text source in the stream. Switching the source can
places in the text where it does not disturb the perception of the only be done in places in the text where it does not disturb the
contents. Text from only one source can be presented in real time perception of the contents. Text from only one source at a time
at a time. The delay will therefore vary. The method also has can be presented in real time. The delay will therefore vary. In
other limitations, but is included in this document as a fallback calls where parties take turns properly by ending their entries
method. In calls where parties take turns properly by ending with a new line, the limitations will have limited influence on
their entries with a new line, the limitations will have limited the user experience. When only two parties send text, these two
influence on the user experience. when only two parties send text, will see the text in real time with no delay. Although this
these two will see the text in real time with no delay. This method also has other limitations, it is included in this document
method is specified as a fallback method in this document. as a fallback method.
RTT transport in WebRTC Real-time text transport in WebRTC:
Transport of real-time text in the WebRTC technology is specified [RFC8865] specifies how the WebRTC data channel can be used to
to use the WebRTC data channel in [RFC8865]. That specification transport real-time text. That specification contains a section
contains a section briefly describing its use in multiparty briefly describing its use in multiparty sessions. The focus of
sessions. The focus of this document is RTP transport. this document is RTP transport. Therefore, even if the WebRTC
Therefore, even if the WebRTC transport provides good multiparty transport provides good multiparty performance, it is only
performance, it is just mentioned in this document in relation to mentioned in this document in relation to providing gateways with
providing gateways with multiparty capabilities between RTP and multiparty capabilities between RTP and WebRTC technologies.
WebRTC technologies.
1.3. Intended application 1.3. Intended Application
The method for multiparty real-time text specified in this document The method for multiparty real-time text specified in this document
is primarily intended for use in transmission between mixers and is primarily intended for use in transmissions between mixers and
endpoints in centralized mixing configurations. It is also endpoints in centralized mixing configurations. It is also
applicable between mixers. An often mentioned application is for applicable between mixers. An often-mentioned application is for
emergency service calls with real-time text and voice, where a call emergency service calls with real-time text and voice, where a call
taker wants to make an attended handover of a call to another agent, taker wants to make an attended handover of a call to another agent
and stay to observe the session. Multimedia conference sessions with and stay on the call to observe the session. Multimedia conference
support for participants to contribute in text is another sessions with support for participants to contribute with text is
application. Conferences with central support for speech-to-text another example. Conferences with central support for speech-to-text
conversion is yet another mentioned application. conversion represent yet another example.
In all these applications, normally only one participant at a time In all these applications, normally only one participant at a time
will send long text utterances. In some cases, one other participant will send long text comments. In some cases, one other participant
will occasionally contribute with a longer comment simultaneously. will occasionally contribute with a longer comment simultaneously.
That may also happen in some rare cases when text is interpreted to That may also happen in some rare cases when text is translated to
text in another language in a conference. Apart from these cases, text in another language in a conference. Apart from these cases,
other participants are only expected to contribute with very brief other participants are only expected to contribute with very brief
utterings while others are sending text. comments while others are sending text.
Users expect that the text they send is presented in real-time in a Users expect the text they send to be presented in real time in a
readable way to the other participants even if they send readable way to the other participants even if they send
simultaneously with other users and even when they make brief edit simultaneously with other users and even when they make brief edit
operations of their text by backspacing and correcting their text. operations of their text by backspacing and correcting their text.
Text is supposed to be human generated, by some text input means, Text is supposed to be human generated, by some means of text input,
such as typing on a keyboard or using speech-to-text technology. such as typing on a keyboard or using speech-to-text technology.
Occasional small cut-and-paste operations may appear even if that is Occasional small cut-and-paste operations may appear even if that is
not the initial purpose of real-time text. not the initial purpose of real-time text.
The real-time characteristics of real-time text is essential for the The real-time characteristics of real-time text are essential for the
participants to be able to contribute to a conversation. If the text participants to be able to contribute to a conversation. If the text
is too much delayed from typing a letter to its presentation, then, is delayed too much between the typing of a character and its
in some conference situations, the opportunity to comment will be presentation, then, in some conference situations, the opportunity to
gone and someone else will grab the turn. A delay of more than one comment will be gone and someone else will grab the turn. A delay of
second in such situations is an obstacle for good conversation. more than one second in such situations is an obstacle to good
conversation.
2. Overview of the two specified solutions and selection of method 2. Overview of the Two Specified Solutions and Selection of Method
This section contains a brief introduction of the two methods This section contains a brief introduction of the two methods
specified in this document. specified in this document.
2.1. The RTP-mixer-based solution for multiparty-aware endpoints 2.1. The RTP-Mixer-Based Solution for Multiparty-Aware Endpoints
This method specifies negotiated use of the RFC 4103 format for This method specifies the negotiated use of the formats described in
multiparty transmission in a single RTP stream. The main purpose of RFC 4103, for multiparty transmissions in a single RTP stream. The
this document is to specify a method for true multiparty real-time main purpose of this document is to specify a method for true
text mixing for multiparty-aware endpoints that can be widely multiparty real-time text mixing for multiparty-aware endpoints that
deployed. The RTP-mixer-based method makes use of the current format can be widely deployed. The RTP-mixer-based method makes use of the
for real-time text in [RFC4103]. It is an update of RFC 4103 by a current format for real-time text as provided in [RFC4103]. This
clarification on one way to use it in the multiparty situation. That method updates RFC 4103 by clarifying one way to use it in the
is done by completing a negotiation for this kind of multiparty multiparty situation. That is done by completing a negotiation for
capability and by interleaving packets from different sources. The this kind of multiparty capability and by interleaving packets from
source is indicated in the CSRC element in the RTP packets. Specific different sources. The source is indicated in the CSRC element in
considerations are made to be able to recover text after packet loss. the RTP packets. Specific considerations are made regarding the
ability to recover text after packet loss.
The detailed procedures for the RTP-mixer-based multiparty-aware case The detailed procedures for the RTP-mixer-based multiparty-aware case
are specified in Section 3. are specified in Section 3.
Please use [RFC4103] as reference when reading the specification. Please refer to [RFC4103] when reading this document.
2.2. Mixing for multiparty-unaware endpoints 2.2. Mixing for Multiparty-Unaware Endpoints
A method is also specified in this document for cases when the This document also specifies a method to be used in cases when the
endpoint participating in a multiparty call does not itself implement endpoint participating in a multiparty call does not itself implement
any solution, or not the same, as the mixer. The method requires the any solution or does not implement the same solution as the mixer.
mixer to insert text dividers and readable labels and only send text This method requires the mixer to insert text dividers and readable
from one source at a time until a suitable point appears for source labels and only send text from one source at a time until a suitable
change. This solution is a fallback method with functional point appears for changing the source. This solution is a fallback
limitations. It acts on the presentation level. method with functional limitations. It operates at the presentation
level.
A mixer SHOULD by default format and transmit text to a call A mixer SHOULD by default format and transmit text to a call
participant to be suitable to present on a multiparty-unaware participant so that the text is suitable for presentation on a
endpoint which has not negotiated any method for true multiparty RTT multiparty-unaware endpoint that has not negotiated any method for
handling, but negotiated a "text/red" or "text/t140" format in a true multiparty real-time text handling but has negotiated a "text/
session. This SHOULD be done if nothing else is specified for the red" or "text/t140" format in a session. This SHOULD be done if
application in order to maintain interoperability. Section 4.2 nothing else is specified for the application, in order to maintain
specifies how this mixing is done. interoperability. Section 4.2 specifies how this mixing is done.
2.3. Offer/answer considerations 2.3. Offer/Answer Considerations
RTP Payload for Text Conversation [RFC4103] specifies use of RTP "RTP Payload for Text Conversation" [RFC4103] specifies the use of
[RFC3550], and a redundancy format "text/red" for increased RTP [RFC3550] and a redundancy format ("text/red", as defined in
robustness of real-time text transmission. This document updates [RFC4102]) for increased robustness of real-time text transmission.
[RFC4103] by introducing a capability negotiation for handling This document updates [RFC4103] by introducing a capability
multiparty real-time text, a way to indicate the source of negotiation for handling multiparty real-time text, a way to indicate
transmitted text, and rules for efficient timing of the transmissions the source of transmitted text, and rules for efficient timing of the
interleaved from different sources. transmissions interleaved from different sources.
The capability negotiation for the "RTP-mixer-based multiparty The capability negotiation for the RTP-mixer-based multiparty method
method" is based on use of the SDP media attribute "rtt-mixer". is based on the use of the SDP media attribute "rtt-mixer".
The syntax is as follows: The syntax is as follows:
"a=rtt-mixer"
If any other method for RTP-based multiparty real-time text gets a=rtt-mixer
specified by additional work, it is assumed that it will be
If in the future any other method for RTP-based multiparty real-time
text is specified by additional work, it is assumed that it will be
recognized by some specific SDP feature exchange. recognized by some specific SDP feature exchange.
2.3.1. Initial offer 2.3.1. Initial Offer
A party intending to set up a session and being willing to use the A party that intends to set up a session and is willing to use the
RTP-mixer-based method of this specification for sending or receiving RTP-mixer-based method provided in this specification for sending,
or both sending and receiving real-time text SHALL include the "rtt- receiving, or both sending and receiving real-time text SHALL include
mixer" SDP attribute in the corresponding "text" media section in the the "rtt-mixer" SDP attribute in the corresponding "text" media
initial offer. section in the initial offer.
The party MAY indicate capability for both the RTP-mixer-based method The party MAY indicate its capability regarding both the RTP-mixer-
of this specification and other methods. based method provided in this specification and other methods.
When the offeror has sent the offer including the "rtt-mixer" When the offerer has sent the offer, which includes the "rtt-mixer"
attribute, it MUST be prepared to receive and handle real-time text attribute, it MUST be prepared to receive and handle real-time text
formatted according to both the method for multiparty-aware parties formatted according to both the method for multiparty-aware parties
specified in Section 3 in this specification and two-party formatted specified in Section 3 and two-party formatted real-time text.
real-time text.
2.3.2. Answering the offer 2.3.2. Answering the Offer
A party receiving an offer containing the "rtt-mixer" SDP attribute A party that receives an offer containing the "rtt-mixer" SDP
and being willing to use the RTP-mixer-based method of this attribute and is willing to use the RTP-mixer-based method provided
specification for sending or receiving or both sending and receiving in this specification for sending, receiving, or both sending and
SHALL include the "rtt-mixer" SDP attribute in the corresponding receiving real-time text SHALL include the "rtt-mixer" SDP attribute
"text" media section in the answer. in the corresponding "text" media section in the answer.
If the offer did not contain the "rtt-mixer" attribute, the answer If the offer did not contain the "rtt-mixer" attribute, the answer
MUST NOT contain the "rtt-mixer" attribute. MUST NOT contain the "rtt-mixer" attribute.
Even when the "rtt-mixer" attribute is successfully negotiated, the Even when the "rtt-mixer" attribute is successfully negotiated, the
parties MAY send and receive two-party coded real-time text. parties MAY send and receive two-party coded real-time text.
An answer MUST NOT include acceptance of more than one method for An answer MUST NOT include acceptance of more than one method for
multiparty real-time text in the same RTP session. multiparty real-time text in the same RTP session.
When the answer including acceptance is transmitted, the answerer When the answer, which includes acceptance, is transmitted, the
MUST be prepared to act on received text in the negotiated session answerer MUST be prepared to act on received text in the negotiated
according to the method for multiparty-aware parties specified in session according to the method for multiparty-aware parties
Section 3 of this specification. Reception of text for a two-party specified in Section 3. Reception of text for a two-party session
session SHALL also be supported. SHALL also be supported.
2.3.3. Offeror processing the answer 2.3.3. Offerer Processing the Answer
When the answer is processed by the offeror, it MUST act as specified When the answer is processed by the offerer, the offerer MUST follow
in Section 2.4 the requirements listed in Section 2.4.
2.3.4. Modifying a session 2.3.4. Modifying a Session
A session MAY be modified at any time by any party offering a A session MAY be modified at any time by any party offering a
modified SDP with or without the "rtt-mixer" SDP attribute expressing modified SDP with or without the "rtt-mixer" SDP attribute expressing
a desired change in the support of multiparty real-time text. a desired change in the support of multiparty real-time text.
If the modified offer adds indication of support for multiparty real- If the modified offer adds the indication of support for multiparty
time text by including the "rtt-mixer" SDP attribute, the procedures real-time text by including the "rtt-mixer" SDP attribute, the
specified in the previous subsections SHALL be applied. procedures specified in the previous subsections SHALL be applied.
If the modified offer deletes indication of support for multiparty If the modified offer deletes the indication of support for
real-time text by excluding the "rtt-mixer" SDP attribute, the answer multiparty real-time text by excluding the "rtt-mixer" SDP attribute,
MUST NOT contain the "rtt-mixer" attribute. After processing this the answer MUST NOT contain the "rtt-mixer" attribute. After
SDP exchange, the parties MUST NOT send real-time text formatted for processing this SDP exchange, the parties MUST NOT send real-time
multiparty-aware parties according to this specification. text formatted for multiparty-aware parties according to this
specification.
2.4. Actions depending on capability negotiation result 2.4. Actions Depending on Capability Negotiation Result
A transmitting party SHALL send text according to the RTP-mixer-based A transmitting party SHALL send text according to the RTP-mixer-based
multiparty method only when the negotiation for that method was multiparty method only when the negotiation for that method was
successful and when it conveys text for another source. In all other successful and when it conveys text for another source. In all other
cases, the packets SHALL be populated and interpreted as for a two- cases, the packets SHALL be populated and interpreted as for a two-
party session. party session.
A party which has negotiated the "rtt-mixer" SDP media attribute MUST A party that has negotiated the "rtt-mixer" SDP media attribute and
populate the CSRC-list, and format the packets according to Section 3 acts as an RTP mixer sending multiparty text MUST (1) populate the
if it acts as an rtp-mixer and sends multiparty text. CSRC list and (2) format the packets according to Section 3.
A party which has negotiated the "rtt-mixer" SDP media attribute MUST A party that has negotiated the "rtt-mixer" SDP media attribute MUST
interpret the contents of the "CC" field, the CSRC-list and the interpret the contents of the CC field, the CSRC list, and the
packets according to Section 3 in received RTP packets in the packets according to Section 3 in received RTP packets in the
corresponding RTP stream. corresponding RTP stream.
A party which has not successfully completed the negotiation of the A party that has not successfully completed the negotiation of the
"rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved "rtt-mixer" SDP media attribute MUST NOT transmit packets interleaved
from different sources in the same RTP stream as specified in from different sources in the same RTP stream, as specified in
Section 3. If the party is a mixer and did declare the "rtt-mixer" Section 3. If the party is a mixer and did declare the "rtt-mixer"
SDP media attribute, it SHOULD perform the procedure for multiparty- SDP media attribute, it SHOULD perform the procedure for multiparty-
unaware endpoints. If the party is not a mixer, it SHOULD transmit unaware endpoints. If the party is not a mixer, it SHOULD transmit
as in a two-party session according to [RFC4103]. as in a two-party session according to [RFC4103].
3. Details for the RTP-mixer-based mixing method for multiparty-aware 3. Details for the RTP-Mixer-Based Mixing Method for Multiparty-Aware
endpoints Endpoints
3.1. Use of fields in the RTP packets 3.1. Use of Fields in the RTP Packets
The CC field SHALL show the number of members in the CSRC list, which The CC field SHALL show the number of members in the CSRC list, which
SHALL be one (1) in transmissions from a mixer when conveying text SHALL be one (1) in transmissions from a mixer when conveying text
from other sources in a multiparty session, and otherwise 0. from other sources in a multiparty session, and otherwise 0.
When text is conveyed by a mixer during a multiparty session, a CSRC When text is conveyed by a mixer during a multiparty session, a CSRC
list SHALL be included in the packet. The single member in the CSRC- list SHALL be included in the packet. The single member in the CSRC
list SHALL contain the SSRC of the source of the T140blocks in the list SHALL contain the SSRC of the source of the T140blocks in the
packet. packet.
When redundancy is used, the RECOMMENDED level of redundancy is to When redundancy is used, the RECOMMENDED level of redundancy is to
use one primary and two redundant generations of T140blocks. In some use one primary and two redundant generations of T140blocks. In some
cases, a primary or redundant T140block is empty, but is still cases, a primary or redundant T140block is empty but is still
represented by a member in the redundancy header. represented by a member in the redundancy header.
In other regards, the contents of the RTP packets are equal to what In other respects, the contents of the RTP packets will be as
is specified in [RFC4103]. specified in [RFC4103].
3.2. Initial transmission of a BOM character 3.2. Initial Transmission of a BOM Character
As soon as a participant is known to participate in a session with As soon as a participant is known to participate in a session with
another entity and is available for text reception, a Unicode Byte- another entity and is available for text reception, a Unicode byte
Order Mark (BOM) character SHALL be sent to it by the other entity order mark (BOM) character SHALL be sent to it by the other entity
according to the procedures in this section. This is useful in many according to the procedures in this section. This is useful in many
configurations to open ports and firewalls and setting up the configurations for opening ports and firewalls and for setting up the
connection between the application and the network. If the connection between the application and the network. If the
transmitter is a mixer, then the source of this character SHALL be transmitter is a mixer, then the source of this character SHALL be
indicated to be the mixer itself. indicated to be the mixer itself.
Note that the BOM character SHALL be transmitted with the same Note that the BOM character SHALL be transmitted with the same
redundancy procedures as any other text. redundancy procedures as any other text.
3.3. Keep-alive 3.3. Keep-Alive
After that, the transmitter SHALL send keep-alive traffic to the After that, the transmitter SHALL send keep-alive traffic to the
receiver(s) at regular intervals when no other traffic has occurred receiver(s) at regular intervals when no other traffic has occurred
during that interval, if that is decided for the actual connection. during that interval, if that is decided upon for the actual
It is RECOMMENDED to use the keep-alive solution from [RFC6263]. The connection. It is RECOMMENDED to use the keep-alive solution
consent check of [RFC7675] is a possible alternative if it is used provided in [RFC6263]. The consent check [RFC7675] is a possible
anyway for other reasons. alternative if it is used anyway for other reasons.
3.4. Transmission interval 3.4. Transmission Interval
A "text/red" or "text/t140" transmitter in a mixer SHALL send packets A "text/red" or "text/t140" transmitter in a mixer SHALL send packets
distributed in time as long as there is something (new or redundant distributed over time as long as there is something (new or redundant
T140blocks) to transmit. The maximum transmission interval between T140blocks) to transmit. The maximum transmission interval between
text transmissions from the same source SHALL then be 330 ms, when no text transmissions from the same source SHALL then be 330 ms, when no
other limitations cause a longer interval to be temporarily used. It other limitations cause a longer interval to be temporarily used. It
is RECOMMENDED to send the next packet to a receiver as soon as new is RECOMMENDED to send the next packet to a receiver as soon as new
text to that receiver is available, as long as the mean character text to that receiver is available, as long as the mean character
rate of new text to the receiver calculated over the last 10 one- rate of new text to the receiver calculated over the last 10 one-
second intervals does not exceed the "cps" value of the receiver. second intervals does not exceed the "cps" value of the receiver.
The intention is to keep the latency low and network load limited The intention is to keep the latency low and network load limited
while keeping good protection against text loss in bursty packet loss while keeping good protection against text loss in bursty packet loss
conditions. The main purpose of the 330 ms interval is for timing of conditions. The main purpose of the 330 ms interval is for the
redundant transmission, when no new text from the same source is timing of redundant transmissions, when no new text from the same
available. source is available.
The reason for the value 330 ms is that many sources of text will The value of 330 ms is used, because many sources of text will
transmit new text with 300 ms intervals during periods of continuous transmit new text at 300 ms intervals during periods of continuous
user typing, and then reception in the mixer of such new text will user typing, and then reception in the mixer of such new text will
cause a combined transmission of the new text and the unsent cause a combined transmission of the new text and the unsent
redundancy from the previous transmission. Only when the user stops redundancy from the previous transmission. Only when the user stops
typing, the 330 ms interval will be applied to send the redundancy. typing will the 330 ms interval be applied to send the redundancy.
If the Characters Per Second (cps) value is reached, a longer If the characters per second ("cps") value is reached, a longer
transmission interval SHALL be applied for text from all sources as transmission interval SHALL be applied for text from all sources as
specified in [RFC4103] and only as much of the text queued for specified in [RFC4103] and only as much of the text queued for
transmission SHALL be sent at the end of each transmission interval transmission SHALL be sent at the end of each transmission interval
as can be allowed without exceeding the "cps" value. Division of as can be allowed without exceeding the "cps" value. Division of
text for partial transmission MUST then be made at T140block borders. text for partial transmission MUST then be made at T140block borders.
When the transmission rate falls under the "cps" value again, the When the transmission rate falls below the "cps" value again, the
transmission intervals SHALL be returned to 330 ms and transmission transmission intervals SHALL be reset to 330 ms and transmission of
of new text SHALL return to be made as soon as new text is available. new text SHALL again be made as soon as new text is available.
NOTE: that extending the transmission intervals during high load | NOTE: Extending the transmission intervals during periods of
periods does not change the number of characters to be conveyed. It | high load does not change the number of characters to be
just evens out the load in time and reduces the number of packets per | conveyed. It just evens out the load over time and reduces the
second. With human created conversational text, the sending user | number of packets per second. With human-created
will eventually take a pause letting transmission catch up. | conversational text, the sending user will eventually take a
| pause, letting transmission catch up.
See also Section 8. See also Section 8.
For a transmitter not acting as a mixer, the transmission interval For a transmitter not acting as a mixer, the transmission interval
principles from [RFC4103] apply, and the normal transmission interval principles provided in [RFC4103] apply, and the normal transmission
SHALL be 300 ms. interval SHALL be 300 ms.
3.5. Only one source per packet 3.5. Only One Source per Packet
New text and redundant copies of earlier text from one source SHALL New text and redundant copies of earlier text from one source SHALL
be transmitted in the same packet if available for transmission at be transmitted in the same packet if available for transmission at
the same time. Text from different sources MUST NOT be transmitted the same time. Text from different sources MUST NOT be transmitted
in the same packet. in the same packet.
3.6. Do not send received text to the originating source 3.6. Do Not Send Received Text to the Originating Source
Text received by a mixer from a participant SHOULD NOT be included in Text received by a mixer from a participant SHOULD NOT be included in
transmission from the mixer to that participant, because the normal transmissions from the mixer to that participant, because for text
behavior of the endpoint is to present locally-produced text locally. that is produced locally, the normal behavior of the endpoint is to
present such text directly when it is produced.
3.7. Clean incoming text 3.7. Clean Incoming Text
A mixer SHALL handle reception, recovery from packet loss, deletion A mixer SHALL handle reception, recovery from packet loss, deletion
of superfluous redundancy, marking of possible text loss and deletion of superfluous redundancy, marking of possible text loss, and
of 'BOM' characters from each participant before queueing received deletion of BOM characters from each participant before queueing
text for transmission to receiving participants as specified in received text for transmission to receiving participants as specified
[RFC4103] for single-party sources and Section 3.16 for multiparty in [RFC4103] for single-party sources and Section 3.16 for multiparty
sources (chained mixers). sources (chained mixers).
3.8. Redundant transmission principles 3.8. Principles of Redundant Transmission
A transmitting party using redundancy SHALL send redundant A transmitting party using redundancy SHALL send redundant
repetitions of T140blocks already transmitted in earlier packets. repetitions of T140blocks already transmitted in earlier packets.
The number of redundant generations of T140blocks to include in The number of redundant generations of T140blocks to include in
transmitted packets SHALL be deduced from the SDP negotiation. It transmitted packets SHALL be deduced from the SDP negotiation. It
SHALL be set to the minimum of the number declared by the two parties SHALL be set to the minimum of the number declared by the two parties
negotiating a connection. It is RECOMMENDED to declare and transmit negotiating a connection. It is RECOMMENDED to declare and transmit
one original and two redundant generations of the T140blocks because one original and two redundant generations of the T140blocks, because
that provides good protection against text loss in case of packet this provides good protection against text loss in the case of packet
loss, and low overhead. loss and also provides low overhead.
3.9. Text placement in packets 3.9. Text Placement in Packets
The mixer SHALL compose and transmit an RTP packet to a receiver when The mixer SHALL compose and transmit an RTP packet to a receiver when
one or more of the following conditions have occurred: one or more of the following conditions have occurred:
* The transmission interval is the normal 330 ms and there is newly * The transmission interval is the normal 330 ms (no matter whether
the transmission interval has passed or not), and there is newly
received unsent text available for transmission to that receiver. received unsent text available for transmission to that receiver.
* The current transmission interval has passed and is longer than * The current transmission interval has passed and is longer than
the normal 330 ms and there is newly received unsent text the normal 330 ms, and there is newly received unsent text
available for transmission to that receiver. available for transmission to that receiver.
* The current transmission interval ( normally 330 ms) has passed * The current transmission interval (normally 330 ms) has passed
since already transmitted text was queued for transmission as since already-transmitted text was queued for transmission as
redundant text. redundant text.
The principles from [RFC4103] apply for populating the header, the The principles provided in [RFC4103] apply for populating the header,
redundancy header and the data in the packet with specifics specified the redundancy header, and the data in the packet with specific
here and in the following sections. information, as detailed here and in the following sections.
At the time of transmission, the mixer SHALL populate the RTP packet At the time of transmission, the mixer SHALL populate the RTP packet
with all T140blocks queued for transmission originating from the with all T140blocks queued for transmission originating from the
source in turn for transmission as long as this is not in conflict source selected for transmission as long as this is not in conflict
with the allowed number of characters per second ("cps") or the with the allowed number of characters per second ("cps") or the
maximum packet size. In this way, the latency of the latest received maximum packet size. In this way, the latency of the latest received
text is kept low even in moments of simultaneous transmission from text is kept low even in moments of simultaneous transmission from
many sources. many sources.
Redundant text SHALL also be included, and the assessment of how much Redundant text SHALL also be included, and the assessment of how much
new text can be included within the maximum packet size MUST take new text can be included within the maximum packet size MUST take
into account that the redundancy has priority to be transmitted in into account that the redundancy has priority to be transmitted in
its entirety. See Section 3.4 its entirety. See Section 3.4.
The SSRC of the source SHALL be placed as the only member in the The SSRC of the source SHALL be placed as the only member in the CSRC
CSRC-list. list.
Note: The CSRC-list in an RTP packet only includes the participant | Note: The CSRC list in an RTP packet only includes the
whose text is included in text blocks. It is not the same as the | participant whose text is included in text blocks. It is not
total list of participants in a conference. With audio and video | the same as the total list of participants in a conference.
media, the CSRC-list would often contain all participants who are not | With audio and video media, the CSRC list would often contain
muted whereas text participants that don't type are completely silent | all participants who are not muted, whereas text participants
and thus are not represented in RTP packet CSRC-lists. | that don't type are completely silent and thus are not
| represented in RTP packet CSRC lists.
3.10. Empty T140blocks 3.10. Empty T140blocks
If no unsent T140blocks were available for a source at the time of If no unsent T140blocks were available for a source at the time of
populating a packet, but T140blocks are available which have not yet populating a packet but already-transmitted T140blocks are available
been sent the full intended number of redundant transmissions, then that have not yet been sent the full intended number of redundant
the primary T140block for that source is composed of an empty transmissions, then the primary area in the packet is composed of an
T140block, and populated (without taking up any length) in a packet empty T140block and included (without taking up any length) in the
for transmission. The corresponding SSRC SHALL be placed as usual in packet for transmission. The corresponding SSRC SHALL be placed as
its place in the CSRC-list. usual in its place in the CSRC list.
The first packet in the session, the first after a source switch, and The first packet in the session, the first after a source switch, and
the first after a pause SHALL be populated with the available the first after a pause SHALL be populated with the available
T140blocks for the source in turn to be sent as primary, and empty T140blocks for the source selected to be sent as the primary, and
T140blocks for the agreed number of redundancy generations. empty T140blocks for the agreed-upon number of redundancy
generations.
3.11. Creation of the redundancy 3.11. Creation of the Redundancy
The primary T140block from a source in the latest transmitted packet The primary T140block from a source in the latest transmitted packet
is saved for populating the first redundant T140block for that source is saved for populating the first redundant T140block for that source
in the next transmission of text from that source. The first in the next transmission of text from that source. The first
redundant T140block for that source from the latest transmission is redundant T140block for that source from the latest transmission is
saved for populating the second redundant T140block in the next saved for populating the second redundant T140block in the next
transmission of text from that source. transmission of text from that source.
Usually this is the level of redundancy used. If a higher level of Usually, this is the level of redundancy used. If a higher level of
redundancy is negotiated, then the procedure SHALL be maintained redundancy is negotiated, then the procedure SHALL be continued until
until all available redundant levels of T140blocks are placed in the all available redundant levels of T140blocks are placed in the
packet. If a receiver has negotiated a lower number of "text/red" packet. If a receiver has negotiated a lower number of "text/red"
generations, then that level SHALL be the maximum used by the generations, then that level SHALL be the maximum used by the
transmitter. transmitter.
The T140blocks saved for transmission as redundant data are assigned The T140blocks saved for transmission as redundant data are assigned
a planned transmission time 330 ms after the current time, but SHOULD a planned transmission time of 330 ms after the current time but
be transmitted earlier if new text for the same source gets in turn SHOULD be transmitted earlier if new text for the same source gets
for transmission before that time. selected for transmission before that time.
3.12. Timer offset fields 3.12. Timer Offset Fields
The timestamp offset values SHALL be inserted in the redundancy The timestamp offset values SHALL be inserted in the redundancy
header, with the time offset from the RTP timestamp in the packet header, with the time offset from the RTP timestamp in the packet
when the corresponding T140block was sent as primary. when the corresponding T140block was sent as the primary.
The timestamp offsets are expressed in the same clock tick units as The timestamp offsets are expressed in the same clock tick units as
the RTP timestamp. the RTP timestamp.
The timestamp offset values for empty T140blocks have no relevance The timestamp offset values for empty T140blocks have no relevance
but SHOULD be assigned realistic values. but SHOULD be assigned realistic values.
3.13. Other RTP header fields 3.13. Other RTP Header Fields
The number of members in the CSRC list (0 or 1) SHALL be placed in The number of members in the CSRC list (0 or 1) SHALL be placed in
the "CC" header field. Only mixers place value 1 in the "CC" field. the CC header field. Only mixers place value 1 in the CC field. A
A value of "0" indicates that the source is the transmitting device value of 0 indicates that the source is the transmitting device
itself and that the source is indicated by the SSRC field. This itself and that the source is indicated by the SSRC field. This
value is used by endpoints, and by mixers sending self-sourced data. value is used by endpoints and also by mixers sending self-sourced
data.
The current time SHALL be inserted in the timestamp. The current time SHALL be inserted in the timestamp.
The SSRC header field SHALL contain the SSRC of the RTP session where The SSRC header field SHALL contain the SSRC of the RTP session where
the packet will be transmitted. the packet will be transmitted.
The M-bit SHALL be handled as specified in [RFC4103]. The M-bit SHALL be handled as specified in [RFC4103].
3.14. Pause in transmission 3.14. Pause in Transmission
When there is no new T140block to transmit, and no redundant When there is no new T140block to transmit and no redundant T140block
T140block that has not been retransmitted the intended number of that has not been retransmitted the intended number of times from any
times from any source, the transmission process SHALL be stopped source, the transmission process SHALL be stopped until either new
until either new T140blocks arrive, or a keep-alive method calls for T140blocks arrive or a keep-alive method calls for transmission of
transmission of keep-alive packets. keep-alive packets.
3.15. RTCP considerations 3.15. RTCP Considerations
A mixer SHALL send RTCP reports with SDES, CNAME, and NAME A mixer SHALL send RTCP reports with SDES, CNAME, and NAME
information about the sources in the multiparty call. This makes it information about the sources in the multiparty call. This makes it
possible for participants to compose a suitable label for text from possible for participants to compose a suitable label for text from
each source. each source.
Privacy considerations SHALL be taken when composing these fields. Privacy considerations SHALL be taken when composing these fields.
They contain name and address information that may be sensitive to They contain name and address information that may be considered
transmit in its entirety, e.g., to unauthenticated participants. sensitive if the information is transmitted in its entirety, e.g., to
unauthenticated participants.
3.16. Reception of multiparty contents 3.16. Reception of Multiparty Contents
The "text/red" receiver included in an endpoint with presentation The "text/red" receiver included in an endpoint with presentation
functions will receive RTP packets in the single stream from the functions will receive RTP packets in the single stream from the
mixer, and SHALL distribute the T140blocks for presentation in mixer and SHALL distribute the T140blocks for presentation in
presentation areas for each source. Other receiver roles, such as presentation areas for each source. Other receiver roles, such as
gateways or chained mixers, are also feasible. They require gateways or chained mixers, are also feasible. Whether the stream
considerations if the stream shall just be forwarded, or distributed will only be forwarded or will be distributed based on the different
based on the different sources. sources must be taken into consideration.
3.16.1. Acting on the source of the packet contents 3.16.1. Acting on the Source of the Packet Contents
If the "CC" field value of a received packet is 1, it indicates that If the CC field value of a received packet is 1, it indicates that
the text is conveyed from a source indicated in the single member in the text is conveyed from a source indicated in the single member in
the CSRC-list, and the receiver MUST act on the source according to the CSRC list, and the receiver MUST act on the source according to
its role. If the CC value is 0, the source is indicated in the SSRC its role. If the CC value is 0, the source is indicated in the SSRC
field. field.
3.16.2. Detection and indication of possible text loss 3.16.2. Detection and Indication of Possible Text Loss
The receiver SHALL monitor the RTP sequence numbers of the received The receiver SHALL monitor the RTP sequence numbers of the received
packets for gaps and packets out of order. If a sequence number gap packets for gaps and for packets received out of order. If a
appears and still exists after some defined short time for jitter and sequence number gap appears and still exists after some defined short
reordering resolution, the packets in the gap SHALL be regarded as time for jitter and reordering resolution, the packets in the gap
lost. SHALL be regarded as lost.
If it is known that only one source is active in the RTP session, If it is known that only one source is active in the RTP session,
then it is likely that a gap equal to or larger than the agreed then it is likely that a gap equal to or larger than the agreed-upon
number of redundancy generations (including the primary) causes text number of redundancy generations (including the primary) causes text
loss. In that case, the receiver SHALL create a t140block with a loss. In that case, the receiver SHALL create a T140block with a
marker for possible text loss [T140ad1] and associate it with the marker for possible text loss [T140ad1], associate it with the
source and insert it in the reception buffer for that source. source, and insert it in the reception buffer for that source.
If it is known that more than one source is active in the RTP If it is known that more than one source is active in the RTP
session, then it is not possible in general to evaluate if text was session, then it is not possible in general to evaluate if text was
lost when packets were lost. With two active sources and the lost when packets were lost. With two active sources and the
recommended number of redundancy generations (3), it can take a gap recommended number of redundancy generations (one original and two
of five consecutive lost packets until any text may be lost, but text redundant), it can take a gap of five consecutive lost packets before
loss can also appear if three non-consecutive packets are lost when any text may be lost, but text loss can also appear if three non-
they contained consecutive data from the same source. A simple consecutive packets are lost when they contained consecutive data
method to decide when there is risk for resulting text loss is to from the same source. A simple method for deciding when there is a
evaluate if three or more packets were lost within one second. If risk of resulting text loss is to evaluate if three or more packets
this simple method is used, then a t140block SHOULD be created with a were lost within one second. If this simple method is used, then a
marker for possible text loss [T140ad1] and associated with the SSRC T140block SHOULD be created with a marker for possible text loss
of the RTP session as a general input from the mixer. [T140ad1] and associated with the SSRC of the RTP session as a
general input from the mixer.
Implementations MAY apply more refined methods for more reliable Implementations MAY apply more refined methods for more reliable
detection of whether text was lost or not. Any refined method SHOULD detection of whether text was lost or not. Any refined method SHOULD
prefer marking possible loss rather than not marking when it is prefer marking possible loss rather than not marking when it is
uncertain if there was loss. uncertain if there was loss.
3.16.3. Extracting text and handling recovery 3.16.3. Extracting Text and Handling Recovery
When applying the following procedures, the effects MUST be When applying the following procedures, the effects of possible
considered of possible timestamp wrap around and the RTP session timestamp wraparound and the RTP session possibly changing the SSRC
possibly changing SSRC. MUST be considered.
When a packet is received in an RTP session using the packetization When a packet is received in an RTP session using the packetization
for multiparty-aware endpoints, its T140blocks SHALL be extracted in for multiparty-aware endpoints, its T140blocks SHALL be extracted as
the following way. described below.
The source SHALL be extracted from the CSRC-list if available, The source SHALL be extracted from the CSRC list if available, and
otherwise from the SSRC. otherwise from the SSRC.
If the received packet is the first packet received from the source, If the received packet is the first packet received from the source,
then all T140blocks in the packet SHALL be retrieved and assigned to then all T140blocks in the packet SHALL be retrieved and assigned to
a receive buffer for the source beginning with the oldest available a receive buffer for that source, beginning with the oldest available
redundant generation, continuing with the younger redundant redundant generation, continuing with the younger redundant
generations in age order and finally the primary. generations in age order, and finally ending with the primary.
Note: The normal case is that in the first packet, only the primary | Note: The normal case is that in the first packet, only the
data has contents. The redundant data has contents in the first | primary data has contents. The redundant data has contents in
received packet from a source only after initial packet loss. | the first received packet from a source only after initial
| packet loss.
If the packet is not the first packet from a source, then if If the packet is not the first packet from a source, then if
redundant data is available, the process SHALL start with the oldest redundant data is available, the process SHALL start with the oldest
generation. The timestamp of that redundant data SHALL be created by generation. The timestamp of that redundant data SHALL be created by
subtracting its timestamp offset from the RTP timestamp. If the subtracting its timestamp offset from the RTP timestamp. If the
resulting timestamp is later than the latest retrieved data from the resulting timestamp is later than the latest retrieved data from the
same source, then the redundant data SHALL be retrieved and appended same source, then the redundant data SHALL be retrieved and appended
to the receive buffer. The process SHALL be continued in the same to the receive buffer. The process SHALL be continued in the same
way for all younger generations of redundant data. After that, the way for all younger generations of redundant data. After that, the
timestamp of the packet SHALL be compared with the timestamp of the timestamp of the packet SHALL be compared with the timestamp of the
latest retrieved data from the same source and if it is later, then latest retrieved data from the same source and if it is later, then
the primary data SHALL be retrieved from the packet and appended to the primary data SHALL be retrieved from the packet and appended to
the receive buffer for the source. the receive buffer for the source.
3.16.4. Delete 'BOM' 3.16.4. Delete BOM
Unicode character 'BOM' is used as a start indication and sometimes The Unicode BOM character is used as a start indication and is
used as a filler or keep alive by transmission implementations. sometimes used as a filler or keep-alive by transmission
These SHALL be deleted after extraction from received packets. implementations. Any BOM characters SHALL be deleted after
extraction from received packets.
3.17. Performance considerations 3.17. Performance Considerations
This solution has good performance with low text delays, as long as This solution has good performance with low text delays, as long as
the mean number of characters per second sent during any 10-second the mean number of characters per second sent during any 10-second
interval from a number of simultaneously sending participants to a interval from a number of simultaneously sending participants to a
receiving participant, does not reach the "cps" value. At higher receiving participant does not reach the "cps" value. At higher
numbers of sent characters per second, a jerkiness is visible in the numbers of sent characters per second, a jerkiness is visible in the
presentation of text. The solution is therefore suitable for presentation of text. The solution is therefore suitable for
emergency service use, relay service use, and small or well-managed emergency service use, relay service use, and small or well-managed
larger multimedia conferences. Only in large unmanaged conferences larger multimedia conferences. In large unmanaged conferences with a
with a high number of participants there may on very rare occasions high number of participants only, on very rare occasions, situations
appear situations when many participants happen to send text might arise where many participants happen to send text
simultaneously. In such circumstances, the result may be simultaneously. In such circumstances, the result may be
unpleasantly jerky presentation of text from each sending unpleasantly jerky presentation of text from each sending
participant. It should be noted that it is only the number of users participant. It should be noted that it is only the number of users
sending text within the same moment that causes jerkiness, not the sending text within the same moment that causes jerkiness, not the
total number of users with RTT capability. total number of users with real-time text capability.
3.18. Security for session control and media 3.18. Security for Session Control and Media
Security mechanisms to provide confidentiality and integrity Security mechanisms to provide confidentiality, integrity protection,
protection and peer authentication SHOULD be applied when possible and peer authentication SHOULD be applied when possible regarding the
regarding the capabilities of the participating devices by use of SIP capabilities of the participating devices by using the Session
over TLS by default according to [RFC5630] section 3.1.3 on the Initiation Protocol (SIP) over TLS by default according to
session control level and by default using DTLS-SRTP [RFC5764] on the Section 3.1.3 of [RFC5630] on the session control level and by
media level. In applications where legacy endpoints without security default using DTLS-SRTP [RFC5764] at the media level. In
are allowed, a negotiation SHOULD be performed to decide if applications where legacy endpoints without security are allowed, a
encryption on the media level will be applied. If no other security negotiation SHOULD be performed to decide if encryption at the media
solution is mandated for the application, then OSRTP [RFC8643] is a level will be applied. If no other security solution is mandated for
suitable method to be applied to negotiate SRTP media security with the application, then the Opportunistic Secure Real-time Transport
DTLS. Most SDP examples below are for simplicity expressed without Protocol (OSRTP) [RFC8643] is a suitable method to be applied to
the security additions. The principles (but not all details) for negotiate SRTP media security with DTLS. For simplicity, most SDP
applying DTLS-SRTP [RFC5764] security are shown in a couple of the examples below are expressed without the security additions. The
following examples. principles (but not all details) for applying DTLS-SRTP security
[RFC5764] are shown in a couple of the following examples.
Further general security considerations are covered in Section 10. Further general security considerations are covered in Section 10.
End-to-end encryption would require further work and could be based End-to-end encryption would require further work and could be based
on WebRTC as specified in Section 1.2 or on double encryption as on WebRTC as specified in Section 1.2 or on double encryption as
specified in [RFC8723]. specified in [RFC8723].
3.19. SDP offer/answer examples 3.19. SDP Offer/Answer Examples
This section shows some examples of SDP for session negotiation of This section shows some examples of SDP for session negotiation of
the real-time text media in SIP sessions. Audio is usually provided the real-time text media in SIP sessions. Audio is usually provided
in the same session, and sometimes also video. The examples only in the same session, and sometimes also video. The examples only
show the part of importance for the real-time text media. The show the part of importance for the real-time text media. The
examples relate to the single RTP stream mixing for multiparty-aware examples relate to the single RTP stream mixing for multiparty-aware
endpoints and for multiparty-unaware endpoints. endpoints and for multiparty-unaware endpoints.
Note: Multiparty RTT MAY also be provided through other methods, | Note: Multiparty real-time text MAY also be provided through
e.g., by a Selective Forwarding Middlebox (SFM). In that case, the | other methods, e.g., by a Selective Forwarding Middlebox (SFM).
SDP of the offer will include something specific for that method, | In that case, the SDP of the offer will include something
e.g., an SDP attribute or another media format. An answer selecting | specific for that method, e.g., an SDP attribute or another
the use of that method would accept it by a corresponding | media format. An answer selecting the use of that method would
acknowledgement included in the SDP. The offer may contain also the | accept it via a corresponding acknowledgement included in the
"rtt-mixer" SDP media attribute for the main RTT media when the | SDP. The offer may also contain the "rtt-mixer" SDP media
offeror has capability for both multiparty methods, while an answer, | attribute for the main real-time text media when the offerer
selecting to use SFM will not include the "rtt-mixer" SDP media | has this capability for both multiparty methods, while an
attribute. | answer, choosing to use SFM, will not include the "rtt-mixer"
| SDP media attribute.
Offer example for "text/red" format and multiparty support: Offer example for the "text/red" format, multiparty support, and
capability for 90 characters per second:
m=text 11000 RTP/AVP 100 98 m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000 a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000 a=fmtp:98 cps=90
a=fmtp:100 98/98/98 a=rtpmap:100 red/1000
a=rtt-mixer a=fmtp:100 98/98/98
a=rtt-mixer
Answer example from a multiparty-aware device Answer example from a multiparty-aware device:
m=text 14000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Offer example for "text/red" format including multiparty m=text 14000 RTP/AVP 100 98
and security: a=rtpmap:98 t140/1000
a=fingerprint: (fingerprint1) a=fmtp:98 cps=90
m=text 11000 RTP/AVP 100 98 a=rtpmap:100 red/1000
a=rtpmap:98 t140/1000 a=fmtp:100 98/98/98
a=rtpmap:100 red/1000 a=rtt-mixer
a=fmtp:100 98/98/98
a=rtt-mixer Offer example for the "text/red" format, including multiparty and
security:
a=fingerprint: (fingerprint1)
m=text 11000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
The "fingerprint" is sufficient to offer DTLS-SRTP, with the media The "fingerprint" is sufficient to offer DTLS-SRTP, with the media
line still indicating RTP/AVP. line still indicating RTP/AVP.
Note: For brevity, the entire value of the SDP fingerprint attribute | Note: For brevity, the entire value of the SDP "fingerprint"
is not shown in this and the following example. | attribute is not shown in this and the following example.
Answer example from a multiparty-aware device with security Answer example from a multiparty-aware device with security:
a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
With the "fingerprint" the device acknowledges use of SRTP/DTLS. a=fingerprint: (fingerprint2)
m=text 16000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
a=rtt-mixer
Answer example from a multiparty-unaware device that also With the "fingerprint", the device acknowledges the use of DTLS-SRTP.
does not support security:
m=text 12000 RTP/AVP 100 98 Answer example from a multiparty-unaware device that also does not
a=rtpmap:98 t140/1000 support security:
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
3.20. Packet sequence example from interleaved transmission m=text 12000 RTP/AVP 100 98
a=rtpmap:98 t140/1000
a=rtpmap:100 red/1000
a=fmtp:100 98/98/98
This example shows a symbolic flow of packets from a mixer including 3.20. Packet Sequence Example from Interleaved Transmission
This example shows a symbolic flow of packets from a mixer, including
loss and recovery. The sequence includes interleaved transmission of loss and recovery. The sequence includes interleaved transmission of
text from two RTT sources A and B. P indicates primary data. R1 is text from two real-time text sources: A and B. P indicates primary
first redundant generation data and R2 is the second redundant data. R1 is the first redundant generation of data, and R2 is the
generation data. A1, B1, A2 etc. are text chunks (T140blocks) second redundant generation of data. A1, B1, A2, etc. are text
received from the respective sources and sent on to the receiver by chunks (T140blocks) received from the respective sources and sent on
the mixer. X indicates a dropped packet between the mixer and a to the receiver by the mixer. X indicates a dropped packet between
receiver. The session is assumed to use original and two redundant the mixer and a receiver. The session is assumed to use the original
generations of RTT. and two redundant generations of real-time text.
|-----------------------| |-----------------------|
|Seq no 101, Time=20400 | |Seq no 101, Time=20400 |
|CC=1 | |CC=1 |
|CSRC list A | |CSRC list A |
|R2: A1, Offset=600 | |R2: A1, Offset=600 |
|R1: A2, Offset=300 | |R1: A2, Offset=300 |
|P: A3 | |P: A3 |
|-----------------------| |-----------------------|
Assuming that earlier packets (with text A1 and A2) were received in Assuming that earlier packets (with text A1 and A2) were received in
sequence, text A3 is received from packet 101 and assigned to sequence, text A3 is received from packet 101 and assigned to
reception buffer A. The mixer is now assumed to have received reception buffer A. The mixer is now assumed to have received
initial text from source B 100 ms after packet 101 and will send that initial text from source B 100 ms after packet 101 and will send that
text. Transmission of A2 and A3 as redundancy is planned for 330 ms text. Transmission of A2 and A3 as redundancy is planned for 330 ms
after packet 101 if no new text from A is ready to be sent before after packet 101 if no new text from A is ready to be sent before
that. that.
|-----------------------| |-----------------------|
|Seq no 102, Time=20500 | |Seq no 102, Time=20500 |
|CC=1 | |CC=1 |
|CSRC list B | |CSRC list B |
|R2 Empty, Offset=600 | |R2 Empty, Offset=600 |
|R1: Empty, Offset=300 | |R1: Empty, Offset=300 |
|P: B1 | |P: B1 |
|-----------------------| |-----------------------|
Packet 102 is received.
B1 is retrieved from this packet. Redundant transmission of
B1 is planned 330 ms after packet 102.
X------------------------| Packet 102 is received.
X Seq no 103, Timer=20730|
X CC=1 |
X CSRC list A |
X R2: A2, Offset=630 |
X R1: A3, Offset=330 |
X P: Empty |
X------------------------|
Packet 103 is assumed to be lost due to network problems.
It contains redundancy for A. Sending A3 as second level
redundancy is planned for 330 ms after packet 103.
X------------------------| B1 is retrieved from this packet. Redundant transmission of B1 is
X Seq no 104, Timer=20800| planned 330 ms after packet 102.
X CC=1 |
X CSRC list B |
X R2: Empty, Offset=600 |
X R1: B1, Offset=300 |
X P: B2 |
X------------------------|
Packet 104 contains text from B, including new B2 and
redundant B1. It is assumed dropped due to network
problems.
The mixer has A3 redundancy to send, but no new text
appears from A and therefore the redundancy is sent
330 ms after the previous packet with text from A.
|------------------------| X------------------------|
| Seq no 105, Timer=21060| X Seq no 103, Timer=20730|
| CC=1 | X CC=1 |
| CSRC list A | X CSRC list A |
| R2: A3, Offset=660 | X R2: A2, Offset=630 |
| R1: Empty, Offset=330 | X R1: A3, Offset=330 |
| P: Empty | X P: Empty |
|------------------------| X------------------------|
Packet 105 is received.
A gap for lost packets 103 and 104 is detected.
Assume that no other loss was detected during the last second.
Then it can be concluded that nothing was totally lost.
R2 is checked. Its original time was 21060-660=20400. Packet 103 is assumed to be lost due to network problems.
A packet with text from A was received with that
timestamp, so nothing needs to be recovered.
B1 and B2 still need to be transmitted as redundancy. It contains redundancy for A. Sending A3 as second-level
This is planned 330 ms after packet 104. That redundancy is planned for 330 ms after packet 103.
would be at 21130.
|-----------------------| X------------------------|
|Seq no 106, Timer=21130| X Seq no 104, Timer=20800|
|CC=1 | X CC=1 |
|CSRC list B | X CSRC list B |
| R2: B1, Offset=630 | X R2: Empty, Offset=600 |
| R1: B2, Offset=330 | X R1: B1, Offset=300 |
| P: Empty | X P: B2 |
|-----------------------| X------------------------|
Packet 106 is received. Packet 104 contains text from B, including new B2 and redundant
B1. It is assumed dropped due to network problems.
The second level redundancy in packet 106 is B1 and has timestamp The mixer has A3 redundancy to send, but no new text appears from
offset 630 ms. The timestamp of packet 106 minus 630 is 20500 which A, and therefore the redundancy is sent 330 ms after the previous
is the timestamp of packet 102 that was received. So B1 does not packet with text from A.
need to be retrieved. The first level redundancy in packet 106 has
offset 330. The timestamp of packet 106 minus 330 is 20800. That is
later than the latest received packet with source B. Therefore B2 is
retrieved and assigned to the input buffer for source B. No primary
is available in packet 106.
After this sequence, A3 and B1 and B2 have been received. In this |------------------------|
case no text was lost. | Seq no 105, Timer=21060|
| CC=1 |
| CSRC list A |
| R2: A3, Offset=660 |
| R1: Empty, Offset=330 |
| P: Empty |
|------------------------|
3.21. Maximum character rate "cps" Packet 105 is received.
The default maximum rate of reception of "text/t140" real-time text A gap for lost packets 103 and 104 is detected. Assume that no
is in [RFC4103] specified to be 30 characters per second. The actual other loss was detected during the last second. It can then be
concluded that nothing was totally lost.
R2 is checked. Its original time was 21060-660=20400. A packet
with text from A was received with that timestamp, so nothing
needs to be recovered.
B1 and B2 still need to be transmitted as redundancy. This is
planned 330 ms after packet 104. That would be at 21130.
|-----------------------|
|Seq no 106, Timer=21130|
|CC=1 |
|CSRC list B |
| R2: B1, Offset=630 |
| R1: B2, Offset=330 |
| P: Empty |
|-----------------------|
Packet 106 is received.
The second-level redundancy in packet 106 is B1 and has a
timestamp offset of 630 ms. The timestamp of packet 106 minus 630
is 20500, which is the timestamp of packet 102 that was received.
So, B1 does not need to be retrieved. The first-level redundancy
in packet 106 has an offset of 330. The timestamp of packet 106
minus 330 is 20800. That is later than the latest received packet
with source B. Therefore, B2 is retrieved and assigned to the
input buffer for source B. No primary is available in packet 106.
After this sequence, A3, B1, and B2 have been received. In this
case, no text was lost.
3.21. Maximum Character Rate "cps" Setting
The default maximum rate of reception of "text/t140" real-time text,
as specified in [RFC4103], is 30 characters per second. The actual
rate is calculated without regard to any redundant text transmission rate is calculated without regard to any redundant text transmission
and is in the multiparty case evaluated for all sources contributing and is, in the multiparty case, evaluated for all sources
to transmission to a receiver. The value MAY be modified in the contributing to transmission to a receiver. The value MAY be
"cps" parameter of the FMTP attribute in the media section for the modified in the "cps" parameter of the "fmtp" attribute for the
"text/t140" media. A mixer combining real-time text from a number of "text/t140" format of the "text" media section.
sources may occasionally have a higher combined flow of text coming
from the sources. Endpoints SHOULD therefore specify a suitable
higher value for the "cps" parameter, corresponding to its real
reception capability. A value for "cps" of 90 SHALL be the default
for the "text/t140" stream in the "text/red" format when multiparty
real-time text is negotiated. See [RFC4103] for the format and use
of the "cps" parameter. The same rules apply for the multiparty case
except for the default value.
4. Presentation level considerations A mixer combining real-time text from a number of sources may
occasionally have a higher combined flow of text coming from the
sources. Endpoints SHOULD therefore include a suitable higher value
for the "cps" parameter, corresponding to its real reception
capability. The default "cps" value 30 can be assumed to be
sufficient for small meetings and well-managed larger conferences
with users only making manual text entry. A "cps" value of 90 can be
assumed to be sufficient even for large unmanaged conferences and for
cases when speech-to-text technologies are used for text entry. This
is also a reachable performance for receivers in modern technologies,
and 90 is therefore the RECOMMENDED "cps" value. See [RFC4103] for
the format and use of the "cps" parameter. The same rules apply for
the multiparty case.
4. Presentation-Level Considerations
"Protocol for multimedia application text conversation" [T140] "Protocol for multimedia application text conversation" [T140]
provides the presentation level requirements for the [RFC4103] provides the presentation-level requirements for RTP transport as
transport. Functions for erasure and other formatting functions are described in [RFC4103]. Functions for erasure and other formatting
specified in [T140] which has the following general statement for the functions are specified in [T140], which has the following general
presentation: statement for the presentation:
"The display of text from the members of the conversation should be | The display of text from the members of the conversation should be
arranged so that the text from each participant is clearly readable, | arranged so that the text from each participant is clearly
and its source and the relative timing of entered text is visualized | readable, and its source and the relative timing of entered text
in the display. Mechanisms for looking back in the contents from the | is visualized in the display. Mechanisms for looking back in the
current session should be provided. The text should be displayed as | contents from the current session should be provided. The text
soon as it is received." | should be displayed as soon as it is received.
Strict application of [T140] is of essence for the interoperability
of real-time text implementations and to fulfill the intention that
the session participants have the same information conveyed in the
text contents of the conversation without necessarily having the
exact same layout of the conversation.
[T140] specifies a set of presentation control codes to include in Strict application of [T140] is essential for the interoperability of
the stream. Some of them are optional. Implementations MUST ignore real-time text implementations and to fulfill the intention that the
optional control codes that they do not support. session participants have the same information conveyed in the text
contents of the conversation without necessarily having the exact
same layout of the conversation.
[T140] specifies a set of presentation control codes (Section 4.2.4)
to include in the stream. Some of them are optional.
Implementations MUST ignore optional control codes that they do not
support.
There is no strict "message" concept in real-time text. The Unicode There is no strict "message" concept in real-time text. The Unicode
Line Separator character SHALL be used as a separator allowing a part Line Separator character SHALL be used as a separator allowing a part
of received text to be grouped in presentation. The characters of received text to be grouped in a presentation. The character
"CRLF" may be used by other implementations as a replacement for Line combination "CRLF" may be used by other implementations as a
Separator. The "CRLF" combination SHALL be erased by just one replacement for the Line Separator. The "CRLF" combination SHALL be
erasing action, the same as the Line Separator. Presentation erased by just one erasing action, the same as the Line Separator.
functions are allowed to group text for presentation in smaller Presentation functions are allowed to group text for presentation in
groups than the line separators imply and present such groups with smaller groups than the Line Separators imply and present such groups
source indication together with text groups from other sources (see with a source indication together with text groups from other sources
the following presentation examples). Erasure has no specific limit (see the following presentation examples). Erasure has no specific
by any delimiter in the text stream. limit by any delimiter in the text stream.
4.1. Presentation by multiparty-aware endpoints 4.1. Presentation by Multiparty-Aware Endpoints
A multiparty-aware receiving party, presenting real-time text MUST A multiparty-aware receiving party presenting real-time text MUST
separate text from different sources and present them in separate separate text from different sources and present them in separate
presentation fields. The receiving party MAY separate presentation presentation fields. The receiving party MAY separate the
of parts of text from a source in readable groups based on other presentation of parts of text from a source in readable groups based
criteria than line separator and merge these groups in the on criteria other than a Line Separator and merge these groups in the
presentation area when it benefits the user to most easily find and presentation area when it benefits the user to most easily find and
read text from the different participants. The criteria MAY e.g., be read text from the different participants. The criteria MAY, for
a received comma, full stop, or other phrase delimiters, or a long example, be a received comma, a full stop, some other type of phrase
pause. delimiter, or a long pause.
When text is received from multiple original sources, the When text is received from multiple original sources, the
presentation SHALL provide a view where text is added in multiple presentation SHALL provide a view where text is added in multiple
presentation fields. presentation fields.
If the presentation presents text from different sources in one If the presentation presents text from different sources in one
common area, the presenting endpoint SHOULD insert text from the common area, the presenting endpoint SHOULD insert text from the
local user ended at suitable points merged with received text to local user, where the text ends at suitable points and is merged
indicate the relative timing for when the text groups were completed. properly with received text to indicate the relative timing for when
In this presentation mode, the receiving endpoint SHALL present the the text groups were completed. In this presentation mode, the
source of the different groups of text. This presentation style is receiving endpoint SHALL present the source of the different groups
called the "chat" style here and provides a possibility to follow of text. This presentation style is called the "chat" style here and
text arriving from multiple parties and the approximate relative time provides the possibility of following text arriving from multiple
that text is received related to text from the local user. parties and the approximate relative time that text is received as
related to text from the local user.
A view of a three-party RTT call in chat style is shown in this A view of a three-party real-time text call in chat style is shown in
example . this example.
_________________________________________________ _________________________________________________
| |^| | |^|
|[Alice] Hi, Alice here. |-| |[Alice] Hi, Alice here. |-|
| | | | | |
|[Bob] Bob as well. | | |[Bob] Bob as well. | |
| | | | | |
|[Eve] Hi, this is Eve, calling from Paris. | | |[Eve] Hi, this is Eve, calling from Paris. | |
| I thought you should be here. | | | I thought you should be here. | |
| | | | | |
|[Alice] I am coming on Thursday, my | | |[Alice] I am coming on Thursday, my | |
| performance is not until Friday morning.| | | performance is not until Friday morning.| |
| | | | | |
|[Bob] And I on Wednesday evening. | | |[Bob] And I on Wednesday evening. | |
| | | | | |
|[Alice] Can we meet on Thursday evening? | | |[Alice] Can we meet on Thursday evening? | |
| | | | | |
|[Eve] Yes, definitely. How about 7pm. | | |[Eve] Yes, definitely. How about 7pm. | |
| at the entrance of the restaurant | | | at the entrance of the restaurant | |
| Le Lion Blanc? | | | Le Lion Blanc? | |
|[Eve] we can have dinner and then take a walk |-| |[Eve] we can have dinner and then take a walk |-|
|______________________________________________|v| |______________________________________________|v|
| <Eve-typing> But I need to be back to |^| | <Eve-typing> But I need to be back to |^|
| the hotel by 11 because I need |-| | the hotel by 11 because I need |-|
| | | | | |
| <Bob-typing> I wou |-| | <Bob-typing> I wou |-|
|______________________________________________|v| |______________________________________________|v|
| of course, I underst | | of course, I underst |
|________________________________________________| |________________________________________________|
Figure 3: Example of a three-party RTT call presented in chat style Figure 1: Example of a Three-Party Real-Time Text Call Presented
seen at participant 'Alice's endpoint. in Chat Style Seen at Participant Alice's Endpoint
Other presentation styles than the chat style MAY be arranged. Presentation styles other than the chat style MAY be arranged.
This figure shows how a coordinated column view MAY be presented. Figure 2 shows how a coordinated column view MAY be presented.
_____________________________________________________________________ _____________________________________________________________________
| Bob | Eve | Alice | | Bob | Eve | Alice |
|____________________|______________________|_______________________| |____________________|______________________|_______________________|
| | |I will arrive by TGV. | | | |I will arrive by TGV. |
|My flight is to Orly| |Convenient to the main | |My flight is to Orly| |Convenient to the main |
| |Hi all, can we plan |station. | | |Hi all, can we plan |station. |
| |for the seminar? | | | |for the seminar? | |
|Eve, will you do | | | |Eve, will you do | | |
|your presentation on| | | |your presentation on| | |
|Friday? |Yes, Friday at 10. | | |Friday? |Yes, Friday at 10. | |
|Fine, wo | |We need to meet befo | |Fine, wo | |We need to meet befo |
|___________________________________________________________________| |___________________________________________________________________|
Figure 4: An example of a coordinated column-view of a three-party Figure 2: An Example of a Coordinated Column View of a
session with entries ordered vertically in approximate time-order. Three-Party Session with Entries Ordered Vertically in
Approximate Time Order
4.2. Multiparty mixing for multiparty-unaware endpoints 4.2. Multiparty Mixing for Multiparty-Unaware Endpoints
When the mixer has indicated RTT multiparty capability in an SDP When the mixer has indicated multiparty real-time text capability in
negotiation, but the multiparty capability negotiation fails with an an SDP negotiation but the multiparty capability negotiation fails
endpoint, then the agreed "text/red" or "text/t140" format SHALL be with an endpoint, the agreed-upon "text/red" or "text/t140" format
used and the mixer SHOULD compose a best-effort presentation of SHALL be used and the mixer SHOULD compose a best-effort presentation
multiparty real-time text in one stream intended to be presented by of multiparty real-time text in one stream intended to be presented
an endpoint with no multiparty awareness, when that is desired in the by an endpoint with no multiparty awareness, when that is desired in
actual implementation. The following specifies a procedure which MAY the actual implementation. The following specifies a procedure that
be applied in that situation. MAY be applied in that situation.
This presentation format has functional limitations and SHOULD be This presentation format has functional limitations and SHOULD be
used only to enable participation in multiparty calls by legacy used only to enable participation in multiparty calls by legacy
deployed endpoints implementing only RFC 4103 without any multiparty deployed endpoints implementing only RFC 4103 without any multiparty
extensions specified in this document. extensions specified in this document.
The principles and procedures below do not specify any new protocol The principles and procedures below do not specify any new protocol
elements. They are instead composed of information from [T140] and elements. They are instead composed of information provided in
an ambition to provide a best-effort presentation on an endpoint [T140] and an ambition to provide a best-effort presentation on an
which has functions originally intended only for two-party calls. endpoint that has functions originally intended only for two-party
calls.
The mixer mixing for multiparty-unaware endpoints SHALL compose a The mixer performing the mixing for multiparty-unaware endpoints
simulated, limited multiparty RTT view suitable for presentation in SHALL compose a simulated, limited multiparty real-time text view
one presentation area. The mixer SHALL group text in suitable groups suitable for presentation in one presentation area. The mixer SHALL
and prepare for presentation of them by inserting a line separator group text in suitable groups and prepare them for presentation by
between them if the transmitted text did not already end with a new inserting a Line Separator between them if the transmitted text did
line (line separator or CRLF). A presentable label SHALL be composed not already end with a new line (Line Separator or CRLF). A
and sent for the source initially in the session and after each presentable label SHALL be composed and sent for the source initially
source switch. With this procedure the time for switching from in the session and after each source switch. With this procedure,
transmission of text from one source to transmission of text from the time for switching from transmission of text from one source to
another source depends on the actions of the users. In order to transmission of text from another source depends on the actions of
expedite source switching, a user can, for example, end its turn with the users. In order to expedite source switching, a user can, for
a new line. example, end its turn with a new line.
4.2.1. Actions by the mixer at reception from the call participants 4.2.1. Actions by the Mixer at Reception from the Call Participants
When text is received by the mixer from the different participants, When text is received by the mixer from the different participants,
the mixer SHALL recover text from redundancy if any packets are lost. the mixer SHALL recover text from redundancy if any packets are lost.
The mark for lost text [T140ad1] SHALL be inserted in the stream if The marker for lost text [T140ad1] SHALL be inserted in the stream if
unrecoverable loss appears. Any Unicode "BOM" characters, possibly unrecoverable loss appears. Any Unicode BOM characters, possibly
used for keep-alive, SHALL be deleted. The time of creation of text used for keep-alives, SHALL be deleted. The time of creation of text
(retrieved from the RTP timestamp) SHALL be stored together with the (retrieved from the RTP timestamp) SHALL be stored together with the
received text from each source in queues for transmission to the received text from each source in queues for transmission to the
recipients in order to be able to evaluate text loss. recipients in order to be able to evaluate text loss.
4.2.2. Actions by the mixer for transmission to the recipients 4.2.2. Actions by the Mixer for Transmission to the Recipients
The following procedure SHALL be applied for each multiparty-unaware The following procedure SHALL be applied for each multiparty-unaware
recipient of multiparty text from the mixer. recipient of multiparty text from the mixer.
The text for transmission SHALL be formatted by the mixer for each The text for transmission SHALL be formatted by the mixer for each
receiving user for presentation in one single presentation area. receiving user for presentation in one single presentation area.
Text received from a participant SHOULD NOT be included in Text received from a participant SHOULD NOT be included in
transmission to that participant because it is usually presented transmissions to that participant, because it is usually presented
locally at transmission time. When there is text available for locally at transmission time. When there is text available for
transmission from the mixer to a receiving party from more than one transmission from the mixer to a receiving party from more than one
participant, the mixer SHALL switch between transmission of text from participant, the mixer SHALL switch between transmission of text from
the different sources at suitable points in the transmitted stream. the different sources at suitable points in the transmitted stream.
When switching source, the mixer SHALL insert a line separator if the When switching the source, the mixer SHALL insert a Line Separator if
already transmitted text did not end with a new line (line separator the already-transmitted text did not end with a new line (Line
or CRLF). A label SHALL be composed of information in the CNAME and Separator or CRLF). A label SHALL be composed of information in the
NAME fields in RTCP reports from the participant to have its text CNAME and NAME fields in RTCP reports from the participant to have
transmitted, or from other session information for that user. The its text transmitted, or from other session information for that
label SHALL be delimited by suitable characters (e.g., '[ ]') and user. The label SHALL be delimited by suitable characters (e.g.,
transmitted. The CSRC SHALL indicate the selected source. Then text "[ ]") and transmitted. The CSRC SHALL indicate the selected source.
from that selected participant SHALL be transmitted until a new Then, text from that selected participant SHALL be transmitted until
suitable point for switching source is reached. a new suitable point for switching the source is reached.
Information available to the mixer for composing the label may Information available to the mixer for composing the label may
contain sensitive personal information that SHOULD NOT be revealed in contain sensitive personal information that SHOULD NOT be revealed in
sessions not securely authenticated and confidentiality protected. sessions not securely authenticated and confidentiality protected.
Privacy considerations regarding how much personal information is Privacy considerations regarding how much personal information is
included in the label SHOULD therefore be taken when composing the included in the label SHOULD therefore be taken when composing the
label. label.
Seeking a suitable point for switching source SHALL be done when Seeking a suitable point for switching the source SHALL be done when
there is older text waiting for transmission from any party than the there is older text waiting for transmission from any party than the
age of the last transmitted text. Suitable points for switching are: age of the last transmitted text. Suitable points for switching are:
* A completed phrase ended by comma * A completed phrase ending with a comma.
* A completed sentence * A completed sentence.
* A new line (line separator or CRLF) * A new line (Line Separator or CRLF).
* A long pause (e.g., > 10 seconds) in received text from the * A long pause (e.g., > 10 seconds) in received text from the
currently transmitted source currently transmitted source.
* If text from one participant has been transmitted with text from * If text from one participant has been transmitted with text from
other sources waiting for transmission for a long time (e.g., > 1 other sources waiting for transmission for a long time (e.g., > 1
minute) and none of the other suitable points for switching has minute) and none of the other suitable points for switching has
occurred, a source switch MAY be forced by the mixer at the next occurred, a source switch MAY be forced by the mixer at the next
word delimiter, and also even if a word delimiter does not occur word delimiter, and also even if a word delimiter does not occur
within a time (e.g., 15 seconds) after the scan for a word within some period of time (e.g., 15 seconds) after the scan for a
delimiter started. word delimiter started.
When switching source, the source which has the oldest text in queue When switching the source, the source that has the oldest text in
SHALL be selected to be transmitted. A character display count SHALL queue SHALL be selected to be transmitted. A character display count
be maintained for the currently transmitted source, starting at zero SHALL be maintained for the currently transmitted source, starting at
after the label is transmitted for the currently transmitted source. zero after the label is transmitted for the currently transmitted
source.
The status SHALL be maintained for the latest control code for Select The status SHALL be maintained for the latest control code for Select
Graphic Rendition (SGR) from each source. If there is an SGR code Graphic Rendition (SGR) from each source. If there is an SGR code
stored as the status for the current source before the source switch stored as the status for the current source before the source switch
is done, a reset of SGR SHALL be sent by the sequence SGR 0 [009B is done, a reset of SGR SHALL be sent by the sequence SGR 0 [U+009B
0000 006D] after the new line and before the new label during a U+0000 U+006D] after the new line and before the new label during a
source switch. See SGR below for an explanation. This transmission source switch. See Section 4.2.4 for an explanation. This
does not influence the display count. transmission does not influence the display count.
If there is an SGR code stored for the new source after the source If there is an SGR code stored for the new source after the source
switch, that SGR code SHALL be transmitted to the recipient before switch, that SGR code SHALL be transmitted to the recipient before
the label. This transmission does not influence the display count. the label. This transmission does not influence the display count.
4.2.3. Actions on transmission of text 4.2.3. Actions on Transmission of Text
Text from a source sent to the recipient SHALL increase the display Text from a source sent to the recipient SHALL increase the display
count by one per transmitted character. count by one per transmitted character.
4.2.4. Actions on transmission of control codes 4.2.4. Actions on Transmission of Control Codes
The following control codes specified by T.140 require specific The following control codes, as specified by T.140 [T140], require
actions. They SHALL cause specific considerations in the mixer. specific actions. They SHALL cause specific considerations in the
Note that the codes presented here are expressed in UCS-16, while mixer. Note that the codes presented here are expressed in UTF-16,
transmission is made in the UTF-8 encoding of these codes. while transmission is made in the UTF-8 encoding of these codes.
BEL 0007 Bell Alert in session. Provides for alerting during an BEL (U+0007): Bell. Alert in session. Provides for alerting during
active session. The display count SHALL NOT be altered. an active session. The display count SHALL NOT be altered.
NEW LINE 2028 Line separator. Check and perform a source switch if NEW LINE (U+2028): Line Separator. Check and perform a source
appropriate. Increase the display count by 1. switch if appropriate. Increase the display count by 1.
CR LF 000D 000A A supported but not preferred way of requesting a CR LF (U+000D U+000A): A supported, but not preferred, way of
new line. Check and perform a source switch if appropriate. requesting a new line. Check and perform a source switch if
Increase the display count by 1. appropriate. Increase the display count by 1.
INT ESC 0061 Interrupt (used to initiate the mode negotiation INT (ESC U+0061): Interrupt (used to initiate the mode negotiation
procedure). The display count SHALL NOT be altered. procedure). The display count SHALL NOT be altered.
SGR 009B Ps 006D Select graphic rendition. Ps is the rendition SGR (U+009B Ps U+006D): Select Graphic Rendition. Ps represents the
parameters specified in ISO 6429. The display count SHALL NOT be rendition parameters specified in [ISO6429]. (For freely
altered. The SGR code SHOULD be stored for the current source. available equivalent information, please see [ECMA-48].) The
display count SHALL NOT be altered. The SGR code SHOULD be stored
for the current source.
SOS 0098 Start of string, used as a general protocol element SOS (U+0098): Start of String. Used as a general protocol element
introducer, followed by a maximum 256-byte string and the ST. The introducer, followed by a maximum 256-byte string and the ST. The
display count SHALL NOT be altered. display count SHALL NOT be altered.
ST 009C String terminator, end of SOS string. The display count ST (U+009C): String Terminator. End of SOS string. The display
SHALL NOT be altered. count SHALL NOT be altered.
ESC 001B Escape - used in control strings. The display count SHALL ESC (U+001B): Escape. Used in control strings. The display count
NOT be altered for the complete escape code. SHALL NOT be altered for the complete escape code.
Byte order mark "BOM" (U+FEFF) "Zero width, no break space", used Byte order mark (BOM) (U+FEFF): "Zero width no-break space". Used
for synchronization and keep-alive. It SHALL be deleted from for synchronization and keep-alive. It SHALL be deleted from
incoming streams. It SHALL also be sent first after session incoming streams. It SHALL also be sent first after session
establishment to the recipient. The display count SHALL NOT be establishment to the recipient. The display count SHALL NOT be
altered. altered.
Missing text mark (U+FFFD) "Replacement character", represented as a Missing text mark (U+FFFD): "Replacement character". Represented as
question mark in a rhombus, or if that is not feasible, replaced a question mark in a rhombus, or, if that is not feasible,
by an apostrophe '. It marks the place in the stream of possible replaced by an apostrophe ('). It marks the place in the stream
text loss. This mark SHALL be inserted by the reception procedure of possible text loss. This mark SHALL be inserted by the
in case of unrecoverable loss of packets. The display count SHALL reception procedure in the case of unrecoverable loss of packets.
be increased by one when sent as for any other character. The display count SHALL be increased by one when sent as for any
other character.
SGR If a control code for selecting graphic rendition (SGR) other SGR: If a control code for SGR other than a reset of the graphic
than reset of the graphic rendition (SGR 0) is sent to a rendition (SGR 0) is sent to a recipient, that control code SHALL
recipient, that control code SHALL also be stored as the status also be stored as the status for the source in the storage for SGR
for the source in the storage for SGR status. If a reset graphic status. If a reset graphic rendition (SGR 0) originating from a
rendition (SGR 0) originating from a source is sent, then the SGR source is sent, then the SGR status storage for that source SHALL
status storage for that source SHALL be cleared. The display be cleared. The display count SHALL NOT be increased.
count SHALL NOT be increased.
BS (U+0008) Back Space, intended to erase the last entered character BS (U+0008): "Back Space". Intended to erase the last entered
by a source. Erasure by backspace cannot always be performed as character by a source. Erasure by backspace cannot always be
the erasing party intended. If an erasing action erases all text performed as the erasing party intended. If an erasing action
up to the end of the leading label after a source switch, then the erases all text up to the end of the leading label after a source
mixer MUST NOT transmit more backspaces. Instead, it is switch, then the mixer MUST NOT transmit more backspaces.
RECOMMENDED that a letter "X" is inserted in the text stream for Instead, it is RECOMMENDED that a letter "X" be inserted in the
each backspace as an indication of the intent to erase more. A text stream for each backspace as an indication of the intent to
new line is usually coded by a Line Separator, but the character erase more. A new line is usually coded by a Line Separator, but
combination "CRLF" MAY be used instead. Erasure of a new line is the character combination "CRLF" MAY be used instead. Erasure of
in both cases done by just one erasing action (Backspace). If the a new line is, in both cases, done by just one erasing action
display count has a positive value it SHALL be decreased by one (backspace). If the display count has a positive value, it SHALL
when the BS is sent. If the display count is at zero, it SHALL be decreased by one when the BS is sent. If the display count is
NOT be altered. at zero, it SHALL NOT be altered.
4.2.5. Packet transmission 4.2.5. Packet Transmission
A mixer transmitting to a multiparty-unaware terminal SHALL send A mixer transmitting to a multiparty-unaware endpoint SHALL send
primary data only from one source per packet. The SSRC SHALL be the primary data only from one source per packet. The SSRC SHALL be the
SSRC of the mixer. The CSRC list SHALL contain one member and be the SSRC of the mixer. The CSRC list MAY contain one member and be the
SSRC of the source of the primary data. SSRC of the source of the primary data.
4.2.6. Functional limitations 4.2.6. Functional Limitations
When a multiparty-unaware endpoint presents a conversation in one When a multiparty-unaware endpoint presents a conversation in one
display area in a chat style, it inserts source indications for display area in a chat style, it inserts source indications for
remote text and local user text as they are merged in completed text remote text and local user text as they are merged in completed text
groups. When an endpoint using this layout receives and presents groups. When an endpoint using this layout receives and presents
text mixed for multiparty-unaware endpoints, there will be two levels text mixed for multiparty-unaware endpoints, there will be two levels
of source indicators for the received text; one generated by the of source indicators for the received text: one generated by the
mixer and inserted in a label after each source switch, and another mixer and inserted in a label after each source switch, and another
generated by the receiving endpoint and inserted after each switch generated by the receiving endpoint and inserted after each switch
between local and remote source in the presentation area. This will between the local source and the remote source in the presentation
waste display space and look inconsistent to the reader. area. This will waste display space and look inconsistent to the
reader.
New text can be presented only from one source at a time. Switch of New text can be presented from only one source at a time. Switching
source to be presented takes place at suitable places in the text, the source to be presented takes place at suitable places in the
such as end of phrase, end of sentence, line separator and text, such as the end of a phrase, the end of a sentence, or a Line
inactivity. Therefore, the time to switch to present waiting text Separator, or upon detecting inactivity. Therefore, the time to
from other sources may become long and will vary and depend on the switch to present waiting text from other sources may grow long, and
actions of the currently presented source. it will vary and depend on the actions of the currently presented
source.
Erasure can only be done up to the latest source switch. If a user Erasure can only be done up to the latest source switch. If a user
tries to erase more text, the erasing actions will be presented as tries to erase more text, the erasing actions will be presented as a
letter X after the label. letter "X" after the label.
Text loss because of network errors may hit the label between entries Text loss because of network errors may hit the label between entries
from different parties, causing risk for misunderstanding from which from different parties, causing the risk of a misunderstanding
source a piece of text is. regarding which source provided a piece of text.
These facts make it strongly RECOMMENDED implementing multiparty Because of these facts, it is strongly RECOMMENDED that multiparty
awareness in RTT endpoints. The use of the mixing method for awareness be implemented in real-time text endpoints. The use of the
multiparty-unaware endpoints should be left for use with endpoints mixing method for multiparty-unaware endpoints should be left for use
which are impossible to upgrade to become multiparty-aware. with endpoints that are impossible to upgrade to become multiparty
aware.
4.2.7. Example views of presentation on multiparty-unaware endpoints 4.2.7. Example Views of Presentation on Multiparty-Unaware Endpoints
The following pictures are examples of the view on a participant's The following pictures are examples of the view on a participant's
display for the multiparty-unaware case. display for the multiparty-unaware case.
_________________________________________________ Figure 3 shows how a coordinated column view MAY be presented on
| Conference | Alice | Alice's device in a view with two columns. The mixer inserts labels
|________________________|_________________________|
| |I will arrive by TGV. |
|[Bob]:My flight is to |Convenient to the main |
|Orly. |station. |
|[Eve]:Hi all, can we | |
|plan for the seminar. | |
| | |
|[Bob]:Eve, will you do | |
|your presentation on | |
|Friday? | |
|[Eve]:Yes, Friday at 10.| |
|[Bob]: Fine, wo |We need to meet befo |
|________________________|_________________________|
Figure 5: Alice who has a conference-unaware client is receiving the
multiparty real-time text in a single-stream.
This figure shows how a coordinated column view MAY be presented on
Alice's device in a view with two-columns. The mixer inserts labels
to show how the sources alternate in the column with received text. to show how the sources alternate in the column with received text.
The mixer alternates between the sources at suitable points in the The mixer alternates between the sources at suitable points in the
text exchange so that text entries from each party can be text exchange so that text entries from each party can be
conveniently read. conveniently read.
_________________________________________________ ___________________________________________________
| |^| | Conference | Alice |
|(Alice) Hi, Alice here. |-| |_________________________|_________________________|
| | | | |I will arrive by TGV. |
|(mix)[Bob)] Bob as well. | | |[Bob]: My flight is to |Convenient to the main |
| | | |Orly. |station. |
|[Eve] Hi, this is Eve, calling from Paris | | |[Eve]: Hi all, can we | |
| I thought you should be here. | | |plan for the seminar. | |
| | | | | |
|(Alice) I am coming on Thursday, my | | |[Bob]: Eve, will you do | |
| performance is not until Friday morning.| | |your presentation on | |
| | | |Friday? | |
|(mix)[Bob] And I on Wednesday evening. | | |[Eve]: Yes, Friday at 10.| |
| | | |[Bob]: Fine, wo |We need to meet befo |
|[Eve] we can have dinner and then walk | | |_________________________|_________________________|
| | |
|[Eve] But I need to be back to | |
| the hotel by 11 because I need | |
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 6: An example of a view of the multiparty-unaware presentation Figure 3: Alice, Who Has a Conference-Unaware Client, Is
in chat style. Alice is the local user. Receiving the Multiparty Real-Time Text in a Single Stream
In this view, there is a tradition in receiving applications to In Figure 4, there is a tradition in receiving applications to
include a label showing the source of the text, here shown with include a label showing the source of the text, here shown with
parenthesis "()". The mixer also inserts source labels for the parentheses "()". The mixer also inserts source labels for the
multiparty call participants, here shown with brackets "[]". multiparty call participants, here shown with brackets "[]".
5. Relation to Conference Control _________________________________________________
5.1. Use with SIP centralized conferencing framework | |^|
|(Alice) Hi, Alice here. |-|
| | |
|(mix)[Bob] Bob as well. | |
| | |
|[Eve] Hi, this is Eve, calling from Paris | |
| I thought you should be here. | |
| | |
|(Alice) I am coming on Thursday, my | |
| performance is not until Friday morning.| |
| | |
|(mix)[Bob] And I on Wednesday evening. | |
| | |
|[Eve] we can have dinner and then walk | |
| | |
|[Eve] But I need to be back to | |
| the hotel by 11 because I need | |
| |-|
|______________________________________________|v|
| of course, I underst |
|________________________________________________|
Figure 4: An Example of a View of the Multiparty-Unaware
Presentation in Chat Style, Where Alice Is the Local User
5. Relationship to Conference Control
5.1. Use with SIP Centralized Conferencing Framework
The Session Initiation Protocol (SIP) conferencing framework, mainly The Session Initiation Protocol (SIP) conferencing framework, mainly
specified in [RFC4353], [RFC4579] and [RFC4575] is suitable for specified in [RFC4353], [RFC4579], and [RFC4575], is suitable for
coordinating sessions including multiparty RTT. The RTT stream coordinating sessions, including multiparty real-time text. The
between the mixer and a participant is one and the same during the real-time text stream between the mixer and a participant is one and
conference. Participants get announced by notifications when the same during the conference. Participants get announced by
participants are joining or leaving, and further user information may notifications when participants are joining or leaving, and further
be provided. The SSRC of the text to expect from joined users MAY be user information may be provided. The SSRC of the text to expect
included in a notification. The notifications MAY be used both for from joined users MAY be included in a notification. The
security purposes and for translation to a label for presentation to notifications MAY be used for both security purposes and translation
other users. to a label for presentation to other users.
5.2. Conference control 5.2. Conference Control
In managed conferences, control of the real-time text media SHOULD be In managed conferences, control of the real-time text media SHOULD be
provided in the same way as other for media, e.g., for muting and provided in the same way as for other media, e.g., for muting and
unmuting by the direction attributes in SDP [RFC8866]. unmuting by the direction attributes in SDP [RFC8866].
Note that floor control functions may be of value for RTT users as Note that floor control functions may be of value for real-time text
well as for users of other media in a conference. users as well as for users of other media in a conference.
6. Gateway Considerations 6. Gateway Considerations
6.1. Gateway considerations with Textphones Multiparty real-time text sessions may involve gateways of different
kinds. Gateways involved in setting up sessions SHALL correctly
reflect the multiparty capability or unawareness of the combination
of the gateway and the remote endpoint beyond the gateway.
multiparty RTT sessions may involve gateways of different kinds. 6.1. Gateway Considerations with Textphones
Gateways involved in setting up sessions SHALL correctly reflect the
multiparty capability or unawareness of the combination of the
gateway and the remote endpoint beyond the gateway.
One case that may occur is a gateway to Public Switched Telephone One case that may occur is a gateway to the Public Switched Telephone
Network (PSTN) for communication with textphones (e.g., TTYs). Network (PSTN) for communication with textphones (e.g., TTYs).
Textphones are limited devices with no multiparty awareness, and it Textphones are limited devices with no multiparty awareness, and it
SHOULD therefore be suitable for the gateway to not indicate SHOULD therefore be appropriate for the gateway to not indicate
multiparty awareness for that case. Another solution is that the multiparty awareness for that case. Another solution is that the
gateway indicates multiparty capability towards the mixer, and gateway indicates multiparty capability towards the mixer and
includes the multiparty mixer function for multiparty-unaware includes the multiparty mixer function for multiparty-unaware
endpoints itself. This solution makes it possible to adapt to the endpoints itself. This solution makes it possible to adapt to the
functional limitations of the textphone. functional limitations of the textphone.
More information on gateways to textphones is found in [RFC5194] More information on gateways to textphones is found in [RFC5194].
6.2. Gateway considerations with WebRTC 6.2. Gateway Considerations with WebRTC
Gateway operation to real-time text in WebRTC may also be required. Gateway operation between RTP-mixer-based multiparty real-time text
In WebRTC, RTT is specified in [RFC8865]. and WebRTC-based real-time text may also be required. Real-time text
transport in WebRTC is specified in [RFC8865].
A multiparty bridge may have functionality for communicating by RTT A multiparty bridge may have functionality for communicating via
both in RTP streams with RTT and WebRTC T.140 data channels. Other real-time text in both (1) RTP streams with real-time text and (2)
configurations may consist of a multiparty bridge with either WebRTC T.140 data channels. Other configurations may consist of a
technology for RTT transport and a separate gateway for conversion of multiparty bridge with either technology for real-time text transport
the text communication streams between RTP and T.140 data channel. and a separate gateway for conversion of the text communication
streams between RTP and T.140 data channels.
In WebRTC, it is assumed that for a multiparty session, one T.140 In WebRTC, it is assumed that for a multiparty session, one T.140
data channel is established for each source from a gateway or bridge data channel is established for each source from a gateway or bridge
to each participant. Each participant also has a data channel with a to each participant. Each participant also has a data channel with a
two-way connection with the gateway or bridge. two-way connection with the gateway or bridge.
The T.140 data channel used both ways is for text from the WebRTC A T.140 data channel used for two-way communication is for text from
user and from the bridge or gateway itself to the WebRTC user. The the WebRTC user and from the bridge or gateway itself to the WebRTC
label parameter of this T.140 data channel is used as the NAME field user. The label parameter of this T.140 data channel is used as the
in RTCP to participants on the RTP side. The other T.140 data NAME field in RTCP to participants on the RTP side. The other T.140
channels are only for text from other participants to the WebRTC data channels are only for text from other participants to the WebRTC
user. user.
When a new participant has entered the session with RTP transport of When a new participant has entered the session with RTP transport of
RTT, a new T.140 channel SHOULD be established to WebRTC users with real-time text, a new T.140 data channel SHOULD be established to
the label parameter composed of information from the NAME field in WebRTC users with the label parameter composed of information from
RTCP on the RTP side. the NAME field in RTCP on the RTP side.
When a new participant has entered the multiparty session with RTT When a new participant has entered the multiparty session with real-
transport in a WebRTC T.140 data channel, the new participant SHOULD time text transport in a WebRTC T.140 data channel, the new
be announced by a notification to RTP users. The label parameter participant SHOULD be announced by a notification to RTP users. The
from the WebRTC side SHOULD be used as the NAME RTCP field on the RTP label parameter from the WebRTC side or other suitable information
side, or other available session information. from the session or stream establishment procedure SHOULD be used to
compose the NAME RTCP field on the RTP side.
When a participant on the RTP side is disconnected from the When a participant on the RTP side is disconnected from the
multiparty session, the corresponding T.140 data channel(s) SHOULD be multiparty session, the corresponding T.140 data channel(s) SHOULD be
closed. closed.
When a WebRTC user of T.140 data channels disconnects from the mixer, When a WebRTC user of T.140 data channels disconnects from the mixer,
the corresponding RTP streams or sources in an RTP-mixed stream the corresponding RTP streams or sources in an RTP-mixed stream
SHOULD be closed. SHOULD be closed.
T.140 data channels MAY be opened and closed by negotiation or T.140 data channels MAY be opened and closed by negotiation or
renegotiation of the session or by any other valid means as specified renegotiation of the session, or by any other valid means, as
in section 1 of [RFC8865]. specified in Section 1 of [RFC8865].
7. Updates to RFC 4103 7. Updates to RFC 4103
This document updates [RFC4103] by introducing an SDP media attribute This document updates [RFC4103] by introducing an SDP media
"rtt-mixer" for negotiation of multiparty-mixing capability with the attribute, "rtt-mixer", for negotiation of multiparty-mixing
[RFC4103] format, and by specifying the rules for packets when capability with the format described in [RFC4103] and by specifying
multiparty capability is negotiated and in use. the rules for packets when multiparty capability is negotiated and in
use.
8. Congestion considerations 8. Congestion Considerations
The congestion considerations and recommended actions from [RFC4103] The congestion considerations and recommended actions provided in
are also valid in multiparty situations. [RFC4103] are also valid in multiparty situations.
The time values SHALL then be applied per source of text sent to a The time values SHALL then be applied per source of text sent to a
receiver. receiver.
If the very unlikely situation appears that many participants in a In the very unlikely event that many participants in a conference
conference send text simultaneously for a long period, a delay may send text simultaneously for a long period of time, a delay may build
build up for presentation of text at the receivers if the limitation up for the presentation of text at the receivers if the limitation in
in characters per second ("cps") to be transmitted to the characters per second ("cps") to be transmitted to the participants
participants is exceeded. More delay than 7 seconds can cause is exceeded. A delay of more than 15 seconds can cause confusion in
confusion in the session. It is therefore RECOMMENDED that an RTP- the session. It is therefore RECOMMENDED that an RTP mixer discard
mixer-based mixer discards such text causing excessive delays and such text causing excessive delays and insert a general indication of
inserts a general indication of possible text loss [T140ad1] in the possible text loss [T140ad1] in the session. If the main text
session. If the main text contributor is indicated in any way, the contributor is indicated in any way, the mixer MAY avoid deleting
mixer MAY avoid deleting text from that participant. It should text from that participant. It should, however, be noted that human
however be noted that human creation of text normally contains creation of text normally contains pauses, when the transmission can
pauses, when the transmission can catch up, so that the transmission catch up, so that transmission-overload situations are expected to be
overload situations are expected to be very rare. very rare.
9. IANA Considerations 9. IANA Considerations
9.1. Registration of the "rtt-mixer" SDP media attribute 9.1. Registration of the "rtt-mixer" SDP Media Attribute
[RFC EDITOR NOTE: Please replace all instances of RFCXXXX with the
RFC number of this document.]
IANA is asked to register the new SDP attribute "rtt-mixer". IANA has registered the new SDP attribute "rtt-mixer".
Contact name: IESG Contact name: IESG
Contact email: iesg@ietf.org Contact email: iesg@ietf.org
Attribute name: rtt-mixer Attribute name: rtt-mixer
Attribute semantics: See RFCXXXX Section 2.3 Attribute semantics: See RFC 9071, Section 2.3
Attribute value: none Attribute value: none
Usage level: media Usage level: media
Purpose: Indicate support by mixer and endpoint of multiparty mixing Purpose: To indicate mixer and endpoint support of multiparty mixing
for real-time text transmission, using a common RTP-stream for for real-time text transmission, using a common RTP stream for
transmission of text from a number of sources mixed with one transmission of text from a number of sources mixed with one
source at a time and the source indicated in a single CSRC-list source at a time and where the source is indicated in a single
member. CSRC-list member.
Charset Dependent: no Charset Dependent: no
O/A procedure: See RFCXXXX Section 2.3 O/A procedures: See RFC 9071, Section 2.3
Mux Category: normal Mux Category: NORMAL
Reference: RFCXXXX Reference: RFC 9071
10. Security Considerations 10. Security Considerations
The RTP-mixer model requires the mixer to be allowed to decrypt, The RTP-mixer model requires the mixer to be allowed to decrypt,
pack, and encrypt secured text from the conference participants. pack, and encrypt secured text from conference participants.
Therefore, the mixer needs to be trusted to maintain confidentiality Therefore, the mixer needs to be trusted to maintain confidentiality
and integrity of the RTT data. This situation is similar to the and integrity of the real-time text data. This situation is similar
situation for handling audio and video media in centralized mixers. to the situation for handling audio and video media in centralized
mixers.
The requirement to transfer information about the user in RTCP The requirement to transfer information about the user in RTCP
reports in SDES, CNAME, and NAME fields, and in conference reports in SDES, CNAME, and NAME fields, and in conference
notifications, may have privacy concerns as already stated in RFC notifications, may have privacy concerns, as already stated in RFC
3550 [RFC3550], and may be restricted for privacy reasons. When used 3550 [RFC3550], and may be restricted for privacy reasons. When used
for creation of readable labels in the presentation, the receiving for the creation of readable labels in the presentation, the
user will then get a more symbolic label for the source. receiving user will then get a more symbolic label for the source.
The services available through the RTT mixer may have special The services available through the real-time text mixer may be of
interest for deaf and hard-of-hearing persons. Some users may want special interest to deaf and hard-of-hearing individuals. Some users
to refrain from revealing such characteristics broadly in may want to refrain from revealing such characteristics broadly in
conferences. The design of the conference systems where the mixer is conferences. Conference systems where the mixer is included MAY need
included MAY need to be made with confidentiality of such to be designed with the confidentiality of such characteristics in
characteristics in mind. mind.
Participants with malicious intentions may appear and e.g., disturb Participants with malicious intentions may appear and, for example,
the multiparty session by emitting a continuous flow of text. They disrupt the multiparty session by emitting a continuous flow of text.
may also send text that appears to originate from other participants. They may also send text that appears to originate from other
Counteractions should be to require secure signaling, media and participants. Countermeasures should include requiring secure
authentication, and to provide higher-layer conference functions signaling, media, and authentication, and providing higher-layer
e.g., for blocking, muting, and expelling participants. conference functions, e.g., for blocking, muting, and expelling
participants.
Participants with malicious intentions may also try to disturb the Participants with malicious intentions may also try to disrupt the
presentation by sending incomplete or malformed control codes. presentation by sending incomplete or malformed control codes.
Handling of text from the different sources by the receivers MUST Handling of text from the different sources by the receivers MUST
therefore be well separated so that the effects of such actions only therefore be well separated so that the effects of such actions only
affect text from the source causing the action. affect text from the source causing the action.
Care should be taken that if use of the mixer is allowed for users Care should be taken to avoid the possibility of attacks by
both with and without security procedures, opens for possible attacks unauthenticated call participants, and even eavesdropping and
by both unauthenticated call participants and even eavesdropping and manipulation of content by non-participants, if the use of the mixer
manipulating of content non-participants. is permitted for users both with and without security procedures.
As already stated in Section 3.18, security in media SHOULD be As already stated in Section 3.18, security in media SHOULD be
applied by using DTLS-SRTP [RFC5764] on the media level. applied by using DTLS-SRTP [RFC5764] at the media level.
Further security considerations specific for this application are Further security considerations specific to this application are
specified in Section 3.18. specified in Section 3.18.
11. Change history 11. References
[RFC Editor: Please remove this section prior to publication.]
11.1. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-20
Inclusion of edits as respone to a comment by Benjamin Kaduk in
section 3.16.3 to make the recovery procedure generic.
Added persons to the acknowledgements and moved acknowledgements to
last in the document.
11.2. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-19
Edits because of comments in a review by Francesca Palombini.
Edits because of comments from Benjamin Kaduk.
Proposed to not change anything because of Robert Wilton's comments.
Two added sentences in the security section to meet comments by Roman
Danyliw.
11.3. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-18
Edits of nits as proposed in a review by Lars Eggert.
Edits as response to review by Martin Duke.
11.4. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-17
Actions on Gen-ART review comments.
Actions on SecDir review comments.
11.5. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-16
Improvements in the offer/answer considerations section by adding
subsections for each phase in the negotiation as requested by IANA
expert review.
11.6. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-15
Actions on review comments from Jurgen Schonwalder:
A bit more about congestion situations and that they are expected to
be very rare.
Explanation of differences in security between the conference-aware
and the conference-unaware case added in security section.
Presentation examples with suource labels made less confusing, and
explained.
Reference to T.140 inserted at first mentioning of T.140.
Reference to RFC 8825 inserted to explain WebRTC
Nit in wording in terminology section adjusted.
11.7. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-14
Changes from comments by Murray Kucherawy during AD review.
Many SHOULD in section 4.2 on multiparty-unaware mixing changed to
SHALL, and the whole section instead specified to be optional
depending on the application.
Some SHOULD in section 3 either explained or changed to SHALL.
In order to have explainable conditions behind SHOULDs, the
transmission interval in 3.4 is changed to as soon as text is
available as a main principle. The call participants send with 300
ms interval so that will create realistic load conditions anyway.
11.8. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-13
Changed year to 2021.
Changed reference to draft on RTT in WebRTC to recently published RFC
8865.
Changed label brackets in example from "[]" to "()" to avoid nits
comment.
Changed reference "RFC 4566" to recently published "RFC 8866"
11.9. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-12
Changes according to responses on comments from Brian Rosen in
Avtcore list on 2020-12-05 and -06.
Changes according to responses to comments by Bernard Aboba in
avtcore list 2020-12-06.
Introduction of an optiona RTP multi-stream mixing method for further
study as proposed by Bernard Aboba.
Changes clarifying how to open and close T.140 data channels included
in 6.2 after comments by Lorenzo Miniero.
Changes to satisfy nits check. Some "not" changed to "NOT" in
normative wording combinations. Some lower case normative words
changed to upper case. A normative reference deleted from the
abstract. Two informative documents moved from normative references
to informative references.
11.10. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-11
Timestamps and timestamp offsets added to the packet examples in
section 3.23, and the description corrected.
A number of minor corrections added in sections 3.10 - 3.23.
11.11. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-10
The packet composition was modified for interleaving packets from
different sources.
The packet reception was modified for the new interleaving method.
The packet sequence examples was adjusted for the new interleaving
method.
Modifications according to responses to Brian Rosen of 2020-11-03
11.12. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-09
Changed name on the SDP media attribute to "rtt-mixer"
Restructure of section 2 for balance between aware and unaware cases.
Moved conference control to own section.
Improved clarification of recovery and loss in the packet sequence
example.
A number of editorial corrections and improvements.
11.13. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-08
Deleted the method requiring a new packet format "text/rex" because
of the longer standardization and implementation period it needs.
Focus on use of RFC 4103 text/red format with shorter transmission
interval, and source indicated in CSRC.
11.14. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-07
Added a method based on the "text/red" format and single source per
packet, negotiated by the "rtt-mixer" SDP attribute.
Added reasoning and recommendation about indication of loss.
The highest number of sources in one packet is 15, not 16. Changed.
Added in information on update to RFC 4103 that RFC 4103 explicitly
allows addition of FEC method. The redundancy is a kind of forward
error correction.
11.15. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-06
Improved definitions list format.
The format of the media subtype parameters is made to match the
requirements.
The mapping of media subtype parameters to SDP is included.
The "cps" parameter belongs to the t140 subtype and does not need to
be registered here.
11.16. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-05
nomenclature and editorial improvements
"this document" used consistently to refer to this document.
11.17. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-04
'Redundancy header' renamed to 'data header'.
More clarifications added.
Language and figure number corrections.
11.18. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-03
Mention possible need to mute and raise hands as for other media.
---done ----
Make sure that use in two-party calls is also possible and explained.
- may need more wording -
Clarify the RTT is often used together with other media. --done--
Tell that text mixing is N-1. A users own text is not received in
the mix. -done-
In 3. correct the interval to: A "text/rex" transmitter SHOULD send
packets distributed in time as long as there is something (new or
redundant T140blocks) to transmit. The maximum transmission interval
SHOULD then be 300 ms. It is RECOMMENDED to send a packet to a
receiver as soon as new text to that receiver is available, as long
as the time after the latest sent packet to the same receiver is more
than 150 ms, and also the maximum character rate to the receiver is
not exceeded. The intention is to keep the latency low while keeping
a good protection against text loss in bursty packet loss conditions.
-done-
In 1.3 say that the format is used both ways. -done-
In 13.1 change presentation area to presentation field so that reader
does not think it shall be totally separated. -done-
In Performance and intro, tell the performance in number of
simultaneous sending users and introduced delay 16, 150 vs
requirements 5 vs 500. -done --
Clarify redundancy level per connection. -done-
Timestamp also for the last data header. To make it possible for all
text to have time offset as for transmission from the source. Make
that header equal to the others. -done-
Mixer always use the CSRC list, even for its own BOM. -done-
Combine all talk about transmission interval (300 ms vs when text has
arrived) in section 3 in one paragraph or close to each other. -done-
Documents the goal of good performance with low delay for 5
simultaneous typers in the introduction. -done-
Describe better that only primary text shall be sent on to receivers.
Redundancy and loss must be resolved by the mixer. -done-
11.19. Changes included in draft-ietf-avtcore-multi-party-rtt-mix-02
SDP and better description and visibility of security by OSRTP RFC
8634 needed.
The description of gatewaying to WebRTC extended.
The description of the data header in the packet is improved.
11.20. Changes to draft-ietf-avtcore-multi-party-rtt-mix-01
2,5,6 More efficient format "text/rex" introduced and attribute
a=rtt-mix deleted.
3. Brief about use of OSRTP for security included- More needed.
4. Brief motivation for the solution and why not rtp-translator is
used added to intro.
7. More limitations for the multiparty-unaware mixing method
inserted.
8. Updates to RFC 4102 and 4103 more clearly expressed.
9. Gateway to WebRTC started. More needed.
11.21. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-03
to draft-ietf-avtcore-multi-party-rtt-mix-00
Changed file name to draft-ietf-avtcore-multi-party-rtt-mix-00
Replaced CDATA in IANA registration table with better coding.
Converted to xml2rfc version 3.
11.22. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-02
to -03
Changed company and e-mail of the author.
Changed title to "RTP-mixer formatting of multi-party Real-time text"
to better match contents.
Check and modification where needed of use of RFC 2119 words SHALL
etc.
More about the CC value in sections on transmitters and receivers so
that 1-to-1 sessions do not use the mixer format.
Enhanced section on presentation for multiparty-unaware endpoints
A paragraph recommending cps=150 inserted in the performance section.
11.23. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-01
to -02
In Abstract and 1. Introduction: Introduced wording about regulatory
requirements.
In section 5: The transmission interval is decreased to 100 ms when
there is text from more than one source to transmit.
In section 11 about SDP negotiation, a SHOULD-requirement is
introduced that the mixer should make a mix for multiparty-unaware
endpoints if the negotiation is not successful. And a reference to a
later chapter about it.
The presentation considerations chapter 14 is extended with more
information about presentation on multiparty-aware endpoints, and a
new section on the multiparty-unaware mixing with low functionality
but SHOULD be implemented in mixers. Presentation examples are
added.
A short chapter 15 on gateway considerations is introduced.
Clarification about the text/t140 format included in chapter 10.
This sentence added to the chapter 10 about use without redundancy.
"The text/red format SHOULD be used unless some other protection
against packet loss is utilized, for example a reliable network or
transport."
Note about deviation from RFC 2198 added in chapter 4.
In chapter 9. "Use with SIP centralized conferencing framework" the
following note is inserted: Note: The CSRC-list in an RTP packet only
includes participants whose text is included in one or more text
blocks. It is not the same as the list of participants in a
conference. With audio and video media, the CSRC-list would often
contain all participants who are not muted whereas text participants
that don't type are completely silent and so don't show up in RTP
packet CSRC-lists.
11.24. Changes from draft-hellstrom-avtcore-multi-party-rtt-source-00
to -01
Editorial cleanup.
Changed capability indication from fmtp-parameter to SDP attribute
"rtt-mix".
Swapped order of redundancy elements in the example to match reality.
Increased the SDP negotiation section 11.1. Normative References
12. References [ECMA-48] Ecma International, "ECMA-48: Control functions for coded
character sets", 5th edition, June 1991,
<https://www.ecma-international.org/publications-and-
standards/standards/ecma-48/>.
12.1. Normative References [ISO6429] ISO/IEC, "Information technology - Control functions for
coded character sets", ISO/IEC ISO/IEC 6429:1992, December
1992, <https://www.iso.org/obp/ui/#iso:std:iso-
iec:6429:ed-3:v1:en>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997, DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>. <https://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>. July 2003, <https://www.rfc-editor.org/info/rfc3550>.
skipping to change at page 48, line 36 skipping to change at line 1852
[RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text [RFC8865] Holmberg, C. and G. Hellström, "T.140 Real-Time Text
Conversation over WebRTC Data Channels", RFC 8865, Conversation over WebRTC Data Channels", RFC 8865,
DOI 10.17487/RFC8865, January 2021, DOI 10.17487/RFC8865, January 2021,
<https://www.rfc-editor.org/info/rfc8865>. <https://www.rfc-editor.org/info/rfc8865>.
[RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP: [RFC8866] Begen, A., Kyzivat, P., Perkins, C., and M. Handley, "SDP:
Session Description Protocol", RFC 8866, Session Description Protocol", RFC 8866,
DOI 10.17487/RFC8866, January 2021, DOI 10.17487/RFC8866, January 2021,
<https://www.rfc-editor.org/info/rfc8866>. <https://www.rfc-editor.org/info/rfc8866>.
[T140] ITU-T, "Recommendation ITU-T T.140 (02/1998), Protocol for [T140] ITU-T, "Protocol for multimedia application text
multimedia application text conversation", February 1998, conversation", ITU-T Recommendation T.140, February 1998,
<https://www.itu.int/rec/T-REC-T.140-199802-I/en>. <https://www.itu.int/rec/T-REC-T.140-199802-I/en>.
[T140ad1] ITU-T, "Recommendation ITU-T.140 Addendum 1 - (02/2000), [T140ad1] ITU-T, "Recommendation T.140 Addendum", February 2000,
Protocol for multimedia application text conversation",
February 2000,
<https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>. <https://www.itu.int/rec/T-REC-T.140-200002-I!Add1/en>.
12.2. Informative References 11.2. Informative References
[RFC4353] Rosenberg, J., "A Framework for Conferencing with the [RFC4353] Rosenberg, J., "A Framework for Conferencing with the
Session Initiation Protocol (SIP)", RFC 4353, Session Initiation Protocol (SIP)", RFC 4353,
DOI 10.17487/RFC4353, February 2006, DOI 10.17487/RFC4353, February 2006,
<https://www.rfc-editor.org/info/rfc4353>. <https://www.rfc-editor.org/info/rfc4353>.
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A [RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A
Session Initiation Protocol (SIP) Event Package for Session Initiation Protocol (SIP) Event Package for
Conference State", RFC 4575, DOI 10.17487/RFC4575, August Conference State", RFC 4575, DOI 10.17487/RFC4575, August
2006, <https://www.rfc-editor.org/info/rfc4575>. 2006, <https://www.rfc-editor.org/info/rfc4575>.
skipping to change at page 49, line 43 skipping to change at line 1904
DOI 10.17487/RFC8723, April 2020, DOI 10.17487/RFC8723, April 2020,
<https://www.rfc-editor.org/info/rfc8723>. <https://www.rfc-editor.org/info/rfc8723>.
[RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for [RFC8825] Alvestrand, H., "Overview: Real-Time Protocols for
Browser-Based Applications", RFC 8825, Browser-Based Applications", RFC 8825,
DOI 10.17487/RFC8825, January 2021, DOI 10.17487/RFC8825, January 2021,
<https://www.rfc-editor.org/info/rfc8825>. <https://www.rfc-editor.org/info/rfc8825>.
Acknowledgements Acknowledgements
The author want to thank the following persons for support, reviews The author wants to thank the following persons for support, reviews,
and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw, and valuable comments: Bernard Aboba, Amanda Baber, Roman Danyliw,
Spencer Dawkins, Martin Duke, Lars Eggert, James Hamlin, Benjamin Spencer Dawkins, Martin Duke, Lars Eggert, James Hamlin, Benjamin
Kaduk, Murray Kucherawy, Paul Kyziwat, Jonathan Lennox, Lorenzo Kaduk, Murray Kucherawy, Paul Kyzivat, Jonathan Lennox, Lorenzo
Miniero, Dan Mongrain, Francesca Palombini, Colin Perkins, Brian Miniero, Dan Mongrain, Francesca Palombini, Colin Perkins, Brian
Rosen, Juergen Schoenwaelder, Rich Salz, Robert Wilton, Dale Worley, Rosen, Rich Salz, Jürgen Schönwälder, Robert Wilton, Dale Worley,
Peter Yee and Yong Xin. Yong Xin, and Peter Yee.
Author's Address Author's Address
Gunnar Hellstrom
Gunnar Hellstrom Accessible Communication Gunnar Hellström
SE-13670 Vendelso Gunnar Hellström Accessible Communication
SE-13670 Vendelsö
Sweden Sweden
Email: gunnar.hellstrom@ghaccess.se Email: gunnar.hellstrom@ghaccess.se
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